Add m-esr52 at 52.6.0

342 files changed, 40757 insertions, 0 deletions

diff --git a/dom/media/webaudio/AlignedTArray.h b/dom/media/webaudio/AlignedTArray.h
new file mode 100644
index 0000000000..afd2f1f48c
--- /dev/null
+++ b/dom/media/webaudio/AlignedTArray.h
@@ -0,0 +1,121 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AlignedTArray_h__
+#define AlignedTArray_h__
+
+#include "mozilla/Alignment.h"
+#include "nsTArray.h"
+
+/**
+ * E: element type, must be a POD type.
+ * N: N bytes alignment for the first element, defaults to 32
+ * S: S bytes of inline storage
+  */
+template <typename E, int S, int N = 32>
+class AlignedAutoTArray : private AutoTArray<E, S + N>
+{
+  static_assert((N & (N-1)) == 0, "N must be power of 2");
+  typedef AutoTArray<E, S + N> base_type;
+public:
+  typedef E                                          elem_type;
+  typedef typename base_type::size_type              size_type;
+  typedef typename base_type::index_type             index_type;
+
+  AlignedAutoTArray() {}
+  explicit AlignedAutoTArray(size_type capacity) : base_type(capacity + sExtra) {}
+  elem_type* Elements() { return getAligned(base_type::Elements()); }
+  const elem_type* Elements() const { return getAligned(base_type::Elements()); }
+  elem_type& operator[](index_type i) { return Elements()[i];}
+  const elem_type& operator[](index_type i) const { return Elements()[i]; }
+
+  void SetLength(size_type newLen)
+  {
+    base_type::SetLength(newLen + sExtra);
+  }
+
+  MOZ_MUST_USE
+  bool SetLength(size_type newLen, const mozilla::fallible_t&)
+  {
+    return base_type::SetLength(newLen + sExtra, mozilla::fallible);
+  }
+
+  size_type Length() const {
+    return base_type::Length() <= sExtra ? 0 : base_type::Length() - sExtra;
+  }
+
+  using base_type::ShallowSizeOfExcludingThis;
+  using base_type::ShallowSizeOfIncludingThis;
+
+private:
+  AlignedAutoTArray(const AlignedAutoTArray& other) = delete;
+  void operator=(const AlignedAutoTArray& other) = delete;
+
+  static const size_type sPadding = N <= MOZ_ALIGNOF(E) ? 0 : N - MOZ_ALIGNOF(E);
+  static const size_type sExtra = (sPadding + sizeof(E) - 1) / sizeof(E);
+
+  template <typename U>
+  static U* getAligned(U* p)
+  {
+    return reinterpret_cast<U*>(((uintptr_t)p + N - 1) & ~(N-1));
+  }
+};
+
+/**
+ * E: element type, must be a POD type.
+ * N: N bytes alignment for the first element, defaults to 32
+  */
+template <typename E, int N = 32>
+class AlignedTArray : private nsTArray_Impl<E, nsTArrayInfallibleAllocator>
+{
+  static_assert((N & (N-1)) == 0, "N must be power of 2");
+  typedef nsTArray_Impl<E, nsTArrayInfallibleAllocator> base_type;
+public:
+  typedef E                                          elem_type;
+  typedef typename base_type::size_type              size_type;
+  typedef typename base_type::index_type             index_type;
+
+  AlignedTArray() {}
+  explicit AlignedTArray(size_type capacity) : base_type(capacity + sExtra) {}
+  elem_type* Elements() { return getAligned(base_type::Elements()); }
+  const elem_type* Elements() const { return getAligned(base_type::Elements()); }
+  elem_type& operator[](index_type i) { return Elements()[i];}
+  const elem_type& operator[](index_type i) const { return Elements()[i]; }
+
+  void SetLength(size_type newLen)
+  {
+    base_type::SetLength(newLen + sExtra);
+  }
+
+  MOZ_MUST_USE
+  bool SetLength(size_type newLen, const mozilla::fallible_t&)
+  {
+    return base_type::SetLength(newLen + sExtra, mozilla::fallible);
+  }
+
+  size_type Length() const {
+    return base_type::Length() <= sExtra ? 0 : base_type::Length() - sExtra;
+  }
+
+  using base_type::ShallowSizeOfExcludingThis;
+  using base_type::ShallowSizeOfIncludingThis;
+
+private:
+  AlignedTArray(const AlignedTArray& other) = delete;
+  void operator=(const AlignedTArray& other) = delete;
+
+  static const size_type sPadding = N <= MOZ_ALIGNOF(E) ? 0 : N - MOZ_ALIGNOF(E);
+  static const size_type sExtra = (sPadding + sizeof(E) - 1) / sizeof(E);
+
+  template <typename U>
+  static U* getAligned(U* p)
+  {
+    return reinterpret_cast<U*>(((uintptr_t)p + N - 1) & ~(N-1));
+  }
+};
+
+
+#endif // AlignedTArray_h__
diff --git a/dom/media/webaudio/AlignmentUtils.h b/dom/media/webaudio/AlignmentUtils.h
new file mode 100644
index 0000000000..6b145a8ca6
--- /dev/null
+++ b/dom/media/webaudio/AlignmentUtils.h
@@ -0,0 +1,29 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AlignmentUtils_h__
+#define AlignmentUtils_h__
+
+#define IS_ALIGNED16(ptr) ((((uintptr_t)ptr + 15) & ~0x0F) == (uintptr_t)ptr)
+
+#ifdef DEBUG
+  #define ASSERT_ALIGNED16(ptr)                                                  \
+            MOZ_ASSERT(IS_ALIGNED16(ptr), \
+                       #ptr " has to be aligned to a 16 byte boundary");
+#else
+  #define ASSERT_ALIGNED16(ptr)
+#endif
+
+#ifdef DEBUG
+  #define ASSERT_MULTIPLE16(v) \
+            MOZ_ASSERT(v % 16 == 0, #v " has to be a a multiple of 16");
+#else
+  #define ASSERT_MULTIPLE16(v)
+#endif
+
+#define ALIGNED16(ptr) (float*)(((uintptr_t)ptr + 15) & ~0x0F);
+
+#endif
diff --git a/dom/media/webaudio/AnalyserNode.cpp b/dom/media/webaudio/AnalyserNode.cpp
new file mode 100644
index 0000000000..64c3cf4dae
--- /dev/null
+++ b/dom/media/webaudio/AnalyserNode.cpp
@@ -0,0 +1,387 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mozilla/dom/AnalyserNode.h"
+#include "mozilla/dom/AnalyserNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/PodOperations.h"
+
+namespace mozilla {
+
+static const uint32_t MAX_FFT_SIZE = 32768;
+static const size_t CHUNK_COUNT = MAX_FFT_SIZE >> WEBAUDIO_BLOCK_SIZE_BITS;
+static_assert(MAX_FFT_SIZE == CHUNK_COUNT * WEBAUDIO_BLOCK_SIZE,
+              "MAX_FFT_SIZE must be a multiple of WEBAUDIO_BLOCK_SIZE");
+static_assert((CHUNK_COUNT & (CHUNK_COUNT - 1)) == 0,
+              "CHUNK_COUNT must be power of 2 for remainder behavior");
+
+namespace dom {
+
+NS_IMPL_ISUPPORTS_INHERITED0(AnalyserNode, AudioNode)
+
+class AnalyserNodeEngine final : public AudioNodeEngine
+{
+  class TransferBuffer final : public Runnable
+  {
+  public:
+    TransferBuffer(AudioNodeStream* aStream,
+                   const AudioChunk& aChunk)
+      : mStream(aStream)
+      , mChunk(aChunk)
+    {
+    }
+
+    NS_IMETHOD Run() override
+    {
+      RefPtr<AnalyserNode> node =
+        static_cast<AnalyserNode*>(mStream->Engine()->NodeMainThread());
+      if (node) {
+        node->AppendChunk(mChunk);
+      }
+      return NS_OK;
+    }
+
+  private:
+    RefPtr<AudioNodeStream> mStream;
+    AudioChunk mChunk;
+  };
+
+public:
+  explicit AnalyserNodeEngine(AnalyserNode* aNode)
+    : AudioNodeEngine(aNode)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  virtual void ProcessBlock(AudioNodeStream* aStream,
+                            GraphTime aFrom,
+                            const AudioBlock& aInput,
+                            AudioBlock* aOutput,
+                            bool* aFinished) override
+  {
+    *aOutput = aInput;
+
+    if (aInput.IsNull()) {
+      // If AnalyserNode::mChunks has only null chunks, then there is no need
+      // to send further null chunks.
+      if (mChunksToProcess == 0) {
+        return;
+      }
+
+      --mChunksToProcess;
+      if (mChunksToProcess == 0) {
+        aStream->ScheduleCheckForInactive();
+      }
+
+    } else {
+      // This many null chunks will be required to empty AnalyserNode::mChunks.
+      mChunksToProcess = CHUNK_COUNT;
+    }
+
+    RefPtr<TransferBuffer> transfer =
+      new TransferBuffer(aStream, aInput.AsAudioChunk());
+    NS_DispatchToMainThread(transfer);
+  }
+
+  virtual bool IsActive() const override
+  {
+    return mChunksToProcess != 0;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  uint32_t mChunksToProcess = 0;
+};
+
+AnalyserNode::AnalyserNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              1,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mAnalysisBlock(2048)
+  , mMinDecibels(-100.)
+  , mMaxDecibels(-30.)
+  , mSmoothingTimeConstant(.8)
+{
+  mStream = AudioNodeStream::Create(aContext,
+                                    new AnalyserNodeEngine(this),
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+
+  // Enough chunks must be recorded to handle the case of fftSize being
+  // increased to maximum immediately before getFloatTimeDomainData() is
+  // called, for example.
+  Unused << mChunks.SetLength(CHUNK_COUNT, fallible);
+
+  AllocateBuffer();
+}
+
+size_t
+AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf);
+  amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+AnalyserNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AnalyserNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv)
+{
+  // Disallow values that are not a power of 2 and outside the [32,32768] range
+  if (aValue < 32 ||
+      aValue > MAX_FFT_SIZE ||
+      (aValue & (aValue - 1)) != 0) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+  if (FftSize() != aValue) {
+    mAnalysisBlock.SetFFTSize(aValue);
+    AllocateBuffer();
+  }
+}
+
+void
+AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv)
+{
+  if (aValue >= mMaxDecibels) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+  mMinDecibels = aValue;
+}
+
+void
+AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv)
+{
+  if (aValue <= mMinDecibels) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+  mMaxDecibels = aValue;
+}
+
+void
+AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv)
+{
+  if (aValue < 0 || aValue > 1) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+  mSmoothingTimeConstant = aValue;
+}
+
+void
+AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray)
+{
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  aArray.ComputeLengthAndData();
+
+  float* buffer = aArray.Data();
+  size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
+
+  for (size_t i = 0; i < length; ++i) {
+    buffer[i] = WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels);
+  }
+}
+
+void
+AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray)
+{
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels);
+
+  aArray.ComputeLengthAndData();
+
+  unsigned char* buffer = aArray.Data();
+  size_t length = std::min(size_t(aArray.Length()), mOutputBuffer.Length());
+
+  for (size_t i = 0; i < length; ++i) {
+    const double decibels = WebAudioUtils::ConvertLinearToDecibels(mOutputBuffer[i], mMinDecibels);
+    // scale down the value to the range of [0, UCHAR_MAX]
+    const double scaled = std::max(0.0, std::min(double(UCHAR_MAX),
+                                                 UCHAR_MAX * (decibels - mMinDecibels) * rangeScaleFactor));
+    buffer[i] = static_cast<unsigned char>(scaled);
+  }
+}
+
+void
+AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray)
+{
+  aArray.ComputeLengthAndData();
+
+  float* buffer = aArray.Data();
+  size_t length = std::min(aArray.Length(), FftSize());
+
+  GetTimeDomainData(buffer, length);
+}
+
+void
+AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray)
+{
+  aArray.ComputeLengthAndData();
+
+  size_t length = std::min(aArray.Length(), FftSize());
+
+  AlignedTArray<float> tmpBuffer;
+  if (!tmpBuffer.SetLength(length, fallible)) {
+    return;
+  }
+
+  GetTimeDomainData(tmpBuffer.Elements(), length);
+
+  unsigned char* buffer = aArray.Data();
+  for (size_t i = 0; i < length; ++i) {
+    const float value = tmpBuffer[i];
+    // scale the value to the range of [0, UCHAR_MAX]
+    const float scaled = std::max(0.0f, std::min(float(UCHAR_MAX),
+                                                 128.0f * (value + 1.0f)));
+    buffer[i] = static_cast<unsigned char>(scaled);
+  }
+}
+
+bool
+AnalyserNode::FFTAnalysis()
+{
+  AlignedTArray<float> tmpBuffer;
+  size_t fftSize = FftSize();
+  if (!tmpBuffer.SetLength(fftSize, fallible)) {
+    return false;
+  }
+
+  float* inputBuffer = tmpBuffer.Elements();
+  GetTimeDomainData(inputBuffer, fftSize);
+  ApplyBlackmanWindow(inputBuffer, fftSize);
+  mAnalysisBlock.PerformFFT(inputBuffer);
+
+  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
+  const double magnitudeScale = 1.0 / fftSize;
+
+  for (uint32_t i = 0; i < mOutputBuffer.Length(); ++i) {
+    double scalarMagnitude = NS_hypot(mAnalysisBlock.RealData(i),
+                                      mAnalysisBlock.ImagData(i)) *
+                             magnitudeScale;
+    mOutputBuffer[i] = mSmoothingTimeConstant * mOutputBuffer[i] +
+                       (1.0 - mSmoothingTimeConstant) * scalarMagnitude;
+  }
+
+  return true;
+}
+
+void
+AnalyserNode::ApplyBlackmanWindow(float* aBuffer, uint32_t aSize)
+{
+  double alpha = 0.16;
+  double a0 = 0.5 * (1.0 - alpha);
+  double a1 = 0.5;
+  double a2 = 0.5 * alpha;
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    double x = double(i) / aSize;
+    double window = a0 - a1 * cos(2 * M_PI * x) + a2 * cos(4 * M_PI * x);
+    aBuffer[i] *= window;
+  }
+}
+
+bool
+AnalyserNode::AllocateBuffer()
+{
+  bool result = true;
+  if (mOutputBuffer.Length() != FrequencyBinCount()) {
+    if (!mOutputBuffer.SetLength(FrequencyBinCount(), fallible)) {
+      return false;
+    }
+    memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount());
+  }
+  return result;
+}
+
+void
+AnalyserNode::AppendChunk(const AudioChunk& aChunk)
+{
+  if (mChunks.Length() == 0) {
+    return;
+  }
+
+  ++mCurrentChunk;
+  mChunks[mCurrentChunk & (CHUNK_COUNT - 1)] = aChunk;
+}
+
+// Reads into aData the oldest aLength samples of the fftSize most recent
+// samples.
+void
+AnalyserNode::GetTimeDomainData(float* aData, size_t aLength)
+{
+  size_t fftSize = FftSize();
+  MOZ_ASSERT(aLength <= fftSize);
+
+  if (mChunks.Length() == 0) {
+    PodZero(aData, aLength);
+    return;
+  }
+
+  size_t readChunk =
+    mCurrentChunk - ((fftSize - 1) >> WEBAUDIO_BLOCK_SIZE_BITS);
+  size_t readIndex = (0 - fftSize) & (WEBAUDIO_BLOCK_SIZE - 1);
+  MOZ_ASSERT(readIndex == 0 || readIndex + fftSize == WEBAUDIO_BLOCK_SIZE);
+
+  for (size_t writeIndex = 0; writeIndex < aLength; ) {
+    const AudioChunk& chunk = mChunks[readChunk & (CHUNK_COUNT - 1)];
+    const size_t channelCount = chunk.ChannelCount();
+    size_t copyLength =
+      std::min<size_t>(aLength - writeIndex, WEBAUDIO_BLOCK_SIZE);
+    float* dataOut = &aData[writeIndex];
+
+    if (channelCount == 0) {
+      PodZero(dataOut, copyLength);
+    } else {
+      float scale = chunk.mVolume / channelCount;
+      { // channel 0
+        auto channelData =
+          static_cast<const float*>(chunk.mChannelData[0]) + readIndex;
+        AudioBufferCopyWithScale(channelData, scale, dataOut, copyLength);
+      }
+      for (uint32_t i = 1; i < channelCount; ++i) {
+        auto channelData =
+          static_cast<const float*>(chunk.mChannelData[i]) + readIndex;
+        AudioBufferAddWithScale(channelData, scale, dataOut, copyLength);
+      }
+    }
+
+    readChunk++;
+    writeIndex += copyLength;
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/AnalyserNode.h b/dom/media/webaudio/AnalyserNode.h
new file mode 100644
index 0000000000..7fca5df6fe
--- /dev/null
+++ b/dom/media/webaudio/AnalyserNode.h
@@ -0,0 +1,90 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AnalyserNode_h_
+#define AnalyserNode_h_
+
+#include "AudioNode.h"
+#include "FFTBlock.h"
+#include "AlignedTArray.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class AnalyserNode final : public AudioNode
+{
+public:
+  explicit AnalyserNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+
+  virtual JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void GetFloatFrequencyData(const Float32Array& aArray);
+  void GetByteFrequencyData(const Uint8Array& aArray);
+  void GetFloatTimeDomainData(const Float32Array& aArray);
+  void GetByteTimeDomainData(const Uint8Array& aArray);
+  uint32_t FftSize() const
+  {
+    return mAnalysisBlock.FFTSize();
+  }
+  void SetFftSize(uint32_t aValue, ErrorResult& aRv);
+  uint32_t FrequencyBinCount() const
+  {
+    return FftSize() / 2;
+  }
+  double MinDecibels() const
+  {
+    return mMinDecibels;
+  }
+  void SetMinDecibels(double aValue, ErrorResult& aRv);
+  double MaxDecibels() const
+  {
+    return mMaxDecibels;
+  }
+  void SetMaxDecibels(double aValue, ErrorResult& aRv);
+  double SmoothingTimeConstant() const
+  {
+    return mSmoothingTimeConstant;
+  }
+  void SetSmoothingTimeConstant(double aValue, ErrorResult& aRv);
+
+  virtual const char* NodeType() const override
+  {
+    return "AnalyserNode";
+  }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  ~AnalyserNode() {}
+
+private:
+  friend class AnalyserNodeEngine;
+  void AppendChunk(const AudioChunk& aChunk);
+  bool AllocateBuffer();
+  bool FFTAnalysis();
+  void ApplyBlackmanWindow(float* aBuffer, uint32_t aSize);
+  void GetTimeDomainData(float* aData, size_t aLength);
+
+private:
+  FFTBlock mAnalysisBlock;
+  nsTArray<AudioChunk> mChunks;
+  double mMinDecibels;
+  double mMaxDecibels;
+  double mSmoothingTimeConstant;
+  size_t mCurrentChunk = 0;
+  AlignedTArray<float> mOutputBuffer;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioBlock.cpp b/dom/media/webaudio/AudioBlock.cpp
new file mode 100644
index 0000000000..a8c714019f
--- /dev/null
+++ b/dom/media/webaudio/AudioBlock.cpp
@@ -0,0 +1,166 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioBlock.h"
+#include "AlignmentUtils.h"
+
+namespace mozilla {
+
+/**
+ * Heap-allocated buffer of channels of 128-sample float arrays, with
+ * threadsafe refcounting.  Typically you would allocate one of these, fill it
+ * in, and then treat it as immutable while it's shared.
+ *
+ * Downstream references are accounted specially so that the creator of the
+ * buffer can reuse and modify its contents next iteration if other references
+ * are all downstream temporary references held by AudioBlock.
+ *
+ * We guarantee 16 byte alignment of the channel data.
+ */
+class AudioBlockBuffer final : public ThreadSharedObject {
+public:
+
+  virtual AudioBlockBuffer* AsAudioBlockBuffer() override { return this; };
+
+  float* ChannelData(uint32_t aChannel)
+  {
+    float* base = reinterpret_cast<float*>(((uintptr_t)(this + 1) + 15) & ~0x0F);
+    ASSERT_ALIGNED16(base);
+    return base + aChannel * WEBAUDIO_BLOCK_SIZE;
+  }
+
+  static already_AddRefed<AudioBlockBuffer> Create(uint32_t aChannelCount)
+  {
+    CheckedInt<size_t> size = WEBAUDIO_BLOCK_SIZE;
+    size *= aChannelCount;
+    size *= sizeof(float);
+    size += sizeof(AudioBlockBuffer);
+    size += 15;  //padding for alignment
+    if (!size.isValid()) {
+      MOZ_CRASH();
+    }
+
+    void* m = moz_xmalloc(size.value());
+    RefPtr<AudioBlockBuffer> p = new (m) AudioBlockBuffer();
+    NS_ASSERTION((reinterpret_cast<char*>(p.get() + 1) - reinterpret_cast<char*>(p.get())) % 4 == 0,
+                 "AudioBlockBuffers should be at least 4-byte aligned");
+    return p.forget();
+  }
+
+  // Graph thread only.
+  void DownstreamRefAdded() { ++mDownstreamRefCount; }
+  void DownstreamRefRemoved() {
+    MOZ_ASSERT(mDownstreamRefCount > 0);
+    --mDownstreamRefCount;
+  }
+  // Whether this is shared by any owners that are not downstream.
+  // Called only from owners with a reference that is not a downstream
+  // reference.  Graph thread only.
+  bool HasLastingShares()
+  {
+    // mRefCnt is atomic and so reading its value is defined even when
+    // modifications may happen on other threads.  mDownstreamRefCount is
+    // not modified on any other thread.
+    //
+    // If all other references are downstream references (managed on this, the
+    // graph thread), then other threads are not using this buffer and cannot
+    // add further references.  This method can safely return false.  The
+    // buffer contents can be modified.
+    //
+    // If there are other references that are not downstream references, then
+    // this method will return true.  The buffer will be assumed to be still
+    // in use and so will not be reused.
+    nsrefcnt count = mRefCnt;
+    // This test is strictly less than because the caller has a reference
+    // that is not a downstream reference.
+    MOZ_ASSERT(mDownstreamRefCount < count);
+    return count != mDownstreamRefCount + 1;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  AudioBlockBuffer() {}
+  ~AudioBlockBuffer() override { MOZ_ASSERT(mDownstreamRefCount == 0); }
+
+  nsAutoRefCnt mDownstreamRefCount;
+};
+
+AudioBlock::~AudioBlock()
+{
+  ClearDownstreamMark();
+}
+
+void
+AudioBlock::SetBuffer(ThreadSharedObject* aNewBuffer)
+{
+  if (aNewBuffer == mBuffer) {
+    return;
+  }
+
+  ClearDownstreamMark();
+
+  mBuffer = aNewBuffer;
+
+  if (!aNewBuffer) {
+    return;
+  }
+
+  AudioBlockBuffer* buffer = aNewBuffer->AsAudioBlockBuffer();
+  if (buffer) {
+    buffer->DownstreamRefAdded();
+    mBufferIsDownstreamRef = true;
+  }
+}
+
+void
+AudioBlock::ClearDownstreamMark() {
+  if (mBufferIsDownstreamRef) {
+    mBuffer->AsAudioBlockBuffer()->DownstreamRefRemoved();
+    mBufferIsDownstreamRef = false;
+  }
+}
+
+bool
+AudioBlock::CanWrite() {
+  // If mBufferIsDownstreamRef is set then the buffer is not ours to use.
+  // It may be in use by another node which is not downstream.
+  return !mBufferIsDownstreamRef &&
+    !mBuffer->AsAudioBlockBuffer()->HasLastingShares();
+}
+
+void
+AudioBlock::AllocateChannels(uint32_t aChannelCount)
+{
+  MOZ_ASSERT(mDuration == WEBAUDIO_BLOCK_SIZE);
+
+  if (mBufferIsDownstreamRef) {
+    // This is not our buffer to re-use.
+    ClearDownstreamMark();
+  } else if (mBuffer && ChannelCount() == aChannelCount) {
+    AudioBlockBuffer* buffer = mBuffer->AsAudioBlockBuffer();
+    if (buffer && !buffer->HasLastingShares()) {
+      MOZ_ASSERT(mBufferFormat == AUDIO_FORMAT_FLOAT32);
+      // No need to allocate again.
+      mVolume = 1.0f;
+      return;
+    }
+  }
+
+  RefPtr<AudioBlockBuffer> buffer = AudioBlockBuffer::Create(aChannelCount);
+  mChannelData.SetLength(aChannelCount);
+  for (uint32_t i = 0; i < aChannelCount; ++i) {
+    mChannelData[i] = buffer->ChannelData(i);
+  }
+  mBuffer = buffer.forget();
+  mVolume = 1.0f;
+  mBufferFormat = AUDIO_FORMAT_FLOAT32;
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioBlock.h b/dom/media/webaudio/AudioBlock.h
new file mode 100644
index 0000000000..c9a5bb4009
--- /dev/null
+++ b/dom/media/webaudio/AudioBlock.h
@@ -0,0 +1,136 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef MOZILLA_AUDIOBLOCK_H_
+#define MOZILLA_AUDIOBLOCK_H_
+
+#include "AudioSegment.h"
+
+namespace mozilla {
+
+/**
+ * An AudioChunk whose buffer contents need to be valid only for one
+ * processing block iteration, after which contents can be overwritten if the
+ * buffer has not been passed to longer term storage or to another thread,
+ * which may happen though AsAudioChunk() or AsMutableChunk().
+ *
+ * Use on graph thread only.
+ */
+class AudioBlock : private AudioChunk
+{
+public:
+  AudioBlock() {
+    mDuration = WEBAUDIO_BLOCK_SIZE;
+    mBufferFormat = AUDIO_FORMAT_SILENCE;
+  }
+  // No effort is made in constructors to ensure that mBufferIsDownstreamRef
+  // is set because the block is expected to be a temporary and so the
+  // reference will be released before the next iteration.
+  // The custom copy constructor is required so as not to set
+  // mBufferIsDownstreamRef without notifying AudioBlockBuffer.
+  AudioBlock(const AudioBlock& aBlock) : AudioChunk(aBlock.AsAudioChunk()) {}
+  explicit AudioBlock(const AudioChunk& aChunk)
+    : AudioChunk(aChunk)
+  {
+    MOZ_ASSERT(aChunk.mDuration == WEBAUDIO_BLOCK_SIZE);
+  }
+  ~AudioBlock();
+
+  using AudioChunk::GetDuration;
+  using AudioChunk::IsNull;
+  using AudioChunk::ChannelCount;
+  using AudioChunk::ChannelData;
+  using AudioChunk::SizeOfExcludingThisIfUnshared;
+  using AudioChunk::SizeOfExcludingThis;
+  // mDuration is not exposed.  Use GetDuration().
+  // mBuffer is not exposed.  Use SetBuffer().
+  using AudioChunk::mChannelData;
+  using AudioChunk::mVolume;
+  using AudioChunk::mBufferFormat;
+
+  const AudioChunk& AsAudioChunk() const { return *this; }
+  AudioChunk* AsMutableChunk() {
+    ClearDownstreamMark();
+    return this;
+  }
+
+  /**
+   * Allocates, if necessary, aChannelCount buffers of WEBAUDIO_BLOCK_SIZE float
+   * samples for writing.
+   */
+  void AllocateChannels(uint32_t aChannelCount);
+
+  /**
+   * ChannelFloatsForWrite() should only be used when the buffers have been
+   * created with AllocateChannels().
+   */
+  float* ChannelFloatsForWrite(size_t aChannel)
+  {
+    MOZ_ASSERT(mBufferFormat == AUDIO_FORMAT_FLOAT32);
+    MOZ_ASSERT(CanWrite());
+    return static_cast<float*>(const_cast<void*>(mChannelData[aChannel]));
+  }
+
+  void SetBuffer(ThreadSharedObject* aNewBuffer);
+  void SetNull(StreamTime aDuration) {
+    MOZ_ASSERT(aDuration == WEBAUDIO_BLOCK_SIZE);
+    SetBuffer(nullptr);
+    mChannelData.Clear();
+    mVolume = 1.0f;
+    mBufferFormat = AUDIO_FORMAT_SILENCE;
+  }
+
+  AudioBlock& operator=(const AudioBlock& aBlock) {
+    // Instead of just copying, mBufferIsDownstreamRef must be first cleared
+    // if set.  It is set again for the new mBuffer if possible.  This happens
+    // in SetBuffer().
+    return *this = aBlock.AsAudioChunk();
+  }
+  AudioBlock& operator=(const AudioChunk& aChunk) {
+    MOZ_ASSERT(aChunk.mDuration == WEBAUDIO_BLOCK_SIZE);
+    SetBuffer(aChunk.mBuffer);
+    mChannelData = aChunk.mChannelData;
+    mVolume = aChunk.mVolume;
+    mBufferFormat = aChunk.mBufferFormat;
+    return *this;
+  }
+
+  bool IsMuted() const { return mVolume == 0.0f; }
+
+  bool IsSilentOrSubnormal() const
+  {
+    if (!mBuffer) {
+      return true;
+    }
+
+    for (uint32_t i = 0, length = mChannelData.Length(); i < length; ++i) {
+      const float* channel = static_cast<const float*>(mChannelData[i]);
+      for (StreamTime frame = 0; frame < mDuration; ++frame) {
+        if (fabs(channel[frame]) >= FLT_MIN) {
+          return false;
+        }
+      }
+    }
+
+    return true;
+  }
+
+private:
+  void ClearDownstreamMark();
+  bool CanWrite();
+
+  // mBufferIsDownstreamRef is set only when mBuffer references an
+  // AudioBlockBuffer created in a different AudioBlock.  That can happen when
+  // this AudioBlock is on a node downstream from the node which created the
+  // buffer.  When this is set, the AudioBlockBuffer is notified that this
+  // reference does prevent the upstream node from re-using the buffer next
+  // iteration and modifying its contents.  The AudioBlockBuffer is also
+  // notified when mBuffer releases this reference.
+  bool mBufferIsDownstreamRef = false;
+};
+
+} // namespace mozilla
+
+#endif // MOZILLA_AUDIOBLOCK_H_
diff --git a/dom/media/webaudio/AudioBuffer.cpp b/dom/media/webaudio/AudioBuffer.cpp
new file mode 100644
index 0000000000..cb834f6a59
--- /dev/null
+++ b/dom/media/webaudio/AudioBuffer.cpp
@@ -0,0 +1,421 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioBuffer.h"
+#include "mozilla/dom/AudioBufferBinding.h"
+#include "jsfriendapi.h"
+#include "mozilla/ErrorResult.h"
+#include "AudioSegment.h"
+#include "AudioChannelFormat.h"
+#include "mozilla/PodOperations.h"
+#include "mozilla/CheckedInt.h"
+#include "mozilla/MemoryReporting.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioBuffer)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioBuffer)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mJSChannels)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+  tmp->ClearJSChannels();
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(AudioBuffer)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(AudioBuffer)
+  NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER
+  for (uint32_t i = 0; i < tmp->mJSChannels.Length(); ++i) {
+    NS_IMPL_CYCLE_COLLECTION_TRACE_JS_MEMBER_CALLBACK(mJSChannels[i])
+  }
+NS_IMPL_CYCLE_COLLECTION_TRACE_END
+
+NS_IMPL_CYCLE_COLLECTION_ROOT_NATIVE(AudioBuffer, AddRef)
+NS_IMPL_CYCLE_COLLECTION_UNROOT_NATIVE(AudioBuffer, Release)
+
+/**
+ * AudioBuffers can be shared between AudioContexts, so we need a separate
+ * mechanism to track their memory usage. This thread-safe class keeps track of
+ * all the AudioBuffers, and gets called back by the memory reporting system
+ * when a memory report is needed, reporting how much memory is used by the
+ * buffers backing AudioBuffer objects. */
+class AudioBufferMemoryTracker : public nsIMemoryReporter
+{
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIMEMORYREPORTER
+
+private:
+  AudioBufferMemoryTracker();
+  virtual ~AudioBufferMemoryTracker();
+
+public:
+  /* Those methods can be called on any thread. */
+  static void RegisterAudioBuffer(const AudioBuffer* aAudioBuffer);
+  static void UnregisterAudioBuffer(const AudioBuffer* aAudioBuffer);
+private:
+  static AudioBufferMemoryTracker* GetInstance();
+  /* Those methods must be called with the lock held. */
+  void RegisterAudioBufferInternal(const AudioBuffer* aAudioBuffer);
+  /* Returns the number of buffers still present in the hash table. */
+  uint32_t UnregisterAudioBufferInternal(const AudioBuffer* aAudioBuffer);
+  void Init();
+
+  /* This protects all members of this class. */
+  static StaticMutex sMutex;
+  static StaticRefPtr<AudioBufferMemoryTracker> sSingleton;
+  nsTHashtable<nsPtrHashKey<const AudioBuffer>> mBuffers;
+};
+
+StaticRefPtr<AudioBufferMemoryTracker> AudioBufferMemoryTracker::sSingleton;
+StaticMutex AudioBufferMemoryTracker::sMutex;
+
+NS_IMPL_ISUPPORTS(AudioBufferMemoryTracker, nsIMemoryReporter);
+
+AudioBufferMemoryTracker* AudioBufferMemoryTracker::GetInstance()
+{
+  sMutex.AssertCurrentThreadOwns();
+  if (!sSingleton) {
+    sSingleton = new AudioBufferMemoryTracker();
+    sSingleton->Init();
+  }
+  return sSingleton;
+}
+
+AudioBufferMemoryTracker::AudioBufferMemoryTracker()
+{
+}
+
+void
+AudioBufferMemoryTracker::Init()
+{
+  RegisterWeakMemoryReporter(this);
+}
+
+AudioBufferMemoryTracker::~AudioBufferMemoryTracker()
+{
+  UnregisterWeakMemoryReporter(this);
+}
+
+void
+AudioBufferMemoryTracker::RegisterAudioBuffer(const AudioBuffer* aAudioBuffer)
+{
+  StaticMutexAutoLock lock(sMutex);
+  AudioBufferMemoryTracker* tracker = AudioBufferMemoryTracker::GetInstance();
+  tracker->RegisterAudioBufferInternal(aAudioBuffer);
+}
+
+void
+AudioBufferMemoryTracker::UnregisterAudioBuffer(const AudioBuffer* aAudioBuffer)
+{
+  StaticMutexAutoLock lock(sMutex);
+  AudioBufferMemoryTracker* tracker = AudioBufferMemoryTracker::GetInstance();
+  uint32_t count;
+  count = tracker->UnregisterAudioBufferInternal(aAudioBuffer);
+  if (count == 0) {
+    sSingleton = nullptr;
+  }
+}
+
+void
+AudioBufferMemoryTracker::RegisterAudioBufferInternal(const AudioBuffer* aAudioBuffer)
+{
+  sMutex.AssertCurrentThreadOwns();
+  mBuffers.PutEntry(aAudioBuffer);
+}
+
+uint32_t
+AudioBufferMemoryTracker::UnregisterAudioBufferInternal(const AudioBuffer* aAudioBuffer)
+{
+  sMutex.AssertCurrentThreadOwns();
+  mBuffers.RemoveEntry(aAudioBuffer);
+  return mBuffers.Count();
+}
+
+MOZ_DEFINE_MALLOC_SIZE_OF(AudioBufferMemoryTrackerMallocSizeOf)
+
+NS_IMETHODIMP
+AudioBufferMemoryTracker::CollectReports(nsIHandleReportCallback* aHandleReport,
+                                         nsISupports* aData, bool)
+{
+  size_t amount = 0;
+
+  for (auto iter = mBuffers.Iter(); !iter.Done(); iter.Next()) {
+    amount += iter.Get()->GetKey()->SizeOfIncludingThis(AudioBufferMemoryTrackerMallocSizeOf);
+  }
+
+  MOZ_COLLECT_REPORT(
+    "explicit/webaudio/audiobuffer", KIND_HEAP, UNITS_BYTES, amount,
+    "Memory used by AudioBuffer objects (Web Audio).");
+
+  return NS_OK;
+}
+
+AudioBuffer::AudioBuffer(AudioContext* aContext, uint32_t aNumberOfChannels,
+                         uint32_t aLength, float aSampleRate,
+                         already_AddRefed<ThreadSharedFloatArrayBufferList>
+                           aInitialContents)
+  : mOwnerWindow(do_GetWeakReference(aContext->GetOwner())),
+    mSharedChannels(aInitialContents),
+    mLength(aLength),
+    mSampleRate(aSampleRate)
+{
+  MOZ_ASSERT(!mSharedChannels ||
+             mSharedChannels->GetChannels() == aNumberOfChannels);
+  mJSChannels.SetLength(aNumberOfChannels);
+  mozilla::HoldJSObjects(this);
+  AudioBufferMemoryTracker::RegisterAudioBuffer(this);
+}
+
+AudioBuffer::~AudioBuffer()
+{
+  AudioBufferMemoryTracker::UnregisterAudioBuffer(this);
+  ClearJSChannels();
+  mozilla::DropJSObjects(this);
+}
+
+void
+AudioBuffer::ClearJSChannels()
+{
+  mJSChannels.Clear();
+}
+
+/* static */ already_AddRefed<AudioBuffer>
+AudioBuffer::Create(AudioContext* aContext, uint32_t aNumberOfChannels,
+                    uint32_t aLength, float aSampleRate,
+                    already_AddRefed<ThreadSharedFloatArrayBufferList>
+                      aInitialContents,
+                    ErrorResult& aRv)
+{
+  // Note that a buffer with zero channels is permitted here for the sake of
+  // AudioProcessingEvent, where channel counts must match parameters passed
+  // to createScriptProcessor(), one of which may be zero.
+  if (aSampleRate < WebAudioUtils::MinSampleRate ||
+      aSampleRate > WebAudioUtils::MaxSampleRate ||
+      aNumberOfChannels > WebAudioUtils::MaxChannelCount ||
+      !aLength || aLength > INT32_MAX) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  RefPtr<AudioBuffer> buffer =
+    new AudioBuffer(aContext, aNumberOfChannels, aLength, aSampleRate,
+                    Move(aInitialContents));
+
+  return buffer.forget();
+}
+
+JSObject*
+AudioBuffer::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioBufferBinding::Wrap(aCx, this, aGivenProto);
+}
+
+bool
+AudioBuffer::RestoreJSChannelData(JSContext* aJSContext)
+{
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    if (mJSChannels[i]) {
+      // Already have data in JS array.
+      continue;
+    }
+
+    // The following code first zeroes the array and then copies our data
+    // into it. We could avoid this with additional JS APIs to construct
+    // an array (or ArrayBuffer) containing initial data.
+    JS::Rooted<JSObject*> array(aJSContext,
+                                JS_NewFloat32Array(aJSContext, mLength));
+    if (!array) {
+      return false;
+    }
+    if (mSharedChannels) {
+      // "4. Attach ArrayBuffers containing copies of the data to the
+      // AudioBuffer, to be returned by the next call to getChannelData."
+      const float* data = mSharedChannels->GetData(i);
+      JS::AutoCheckCannotGC nogc;
+      bool isShared;
+      mozilla::PodCopy(JS_GetFloat32ArrayData(array, &isShared, nogc), data, mLength);
+      MOZ_ASSERT(!isShared); // Was created as unshared above
+    }
+    mJSChannels[i] = array;
+  }
+
+  mSharedChannels = nullptr;
+
+  return true;
+}
+
+void
+AudioBuffer::CopyFromChannel(const Float32Array& aDestination, uint32_t aChannelNumber,
+                             uint32_t aStartInChannel, ErrorResult& aRv)
+{
+  aDestination.ComputeLengthAndData();
+
+  uint32_t length = aDestination.Length();
+  CheckedInt<uint32_t> end = aStartInChannel;
+  end += length;
+  if (aChannelNumber >= NumberOfChannels() ||
+      !end.isValid() || end.value() > mLength) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  JS::AutoCheckCannotGC nogc;
+  JSObject* channelArray = mJSChannels[aChannelNumber];
+  const float* sourceData = nullptr;
+  if (channelArray) {
+    if (JS_GetTypedArrayLength(channelArray) != mLength) {
+      // The array's buffer was detached.
+      aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+      return;
+    }
+
+    bool isShared = false;
+    sourceData = JS_GetFloat32ArrayData(channelArray, &isShared, nogc);
+    // The sourceData arrays should all have originated in
+    // RestoreJSChannelData, where they are created unshared.
+    MOZ_ASSERT(!isShared);
+  } else if (mSharedChannels) {
+    sourceData = mSharedChannels->GetData(aChannelNumber);
+  }
+
+  if (sourceData) {
+    PodMove(aDestination.Data(), sourceData + aStartInChannel, length);
+  } else {
+    PodZero(aDestination.Data(), length);
+  }
+}
+
+void
+AudioBuffer::CopyToChannel(JSContext* aJSContext, const Float32Array& aSource,
+                           uint32_t aChannelNumber, uint32_t aStartInChannel,
+                           ErrorResult& aRv)
+{
+  aSource.ComputeLengthAndData();
+
+  uint32_t length = aSource.Length();
+  CheckedInt<uint32_t> end = aStartInChannel;
+  end += length;
+  if (aChannelNumber >= NumberOfChannels() ||
+      !end.isValid() || end.value() > mLength) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  if (!RestoreJSChannelData(aJSContext)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  JS::AutoCheckCannotGC nogc;
+  JSObject* channelArray = mJSChannels[aChannelNumber];
+  if (JS_GetTypedArrayLength(channelArray) != mLength) {
+    // The array's buffer was detached.
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  bool isShared = false;
+  float* channelData = JS_GetFloat32ArrayData(channelArray, &isShared, nogc);
+  // The channelData arrays should all have originated in
+  // RestoreJSChannelData, where they are created unshared.
+  MOZ_ASSERT(!isShared);
+  PodMove(channelData + aStartInChannel, aSource.Data(), length);
+}
+
+void
+AudioBuffer::GetChannelData(JSContext* aJSContext, uint32_t aChannel,
+                            JS::MutableHandle<JSObject*> aRetval,
+                            ErrorResult& aRv)
+{
+  if (aChannel >= NumberOfChannels()) {
+    aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+    return;
+  }
+
+  if (!RestoreJSChannelData(aJSContext)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  aRetval.set(mJSChannels[aChannel]);
+}
+
+already_AddRefed<ThreadSharedFloatArrayBufferList>
+AudioBuffer::StealJSArrayDataIntoSharedChannels(JSContext* aJSContext)
+{
+  // "1. If any of the AudioBuffer's ArrayBuffer have been detached, abort
+  // these steps, and return a zero-length channel data buffers to the
+  // invoker."
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    JSObject* channelArray = mJSChannels[i];
+    if (!channelArray || mLength != JS_GetTypedArrayLength(channelArray)) {
+      // Either empty buffer or one of the arrays' buffers was detached.
+      return nullptr;
+    }
+  }
+
+  // "2. Detach all ArrayBuffers for arrays previously returned by
+  // getChannelData on this AudioBuffer."
+  // "3. Retain the underlying data buffers from those ArrayBuffers and return
+  // references to them to the invoker."
+  RefPtr<ThreadSharedFloatArrayBufferList> result =
+    new ThreadSharedFloatArrayBufferList(mJSChannels.Length());
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    JS::Rooted<JSObject*> arrayBufferView(aJSContext, mJSChannels[i]);
+    bool isSharedMemory;
+    JS::Rooted<JSObject*> arrayBuffer(aJSContext,
+                                      JS_GetArrayBufferViewBuffer(aJSContext,
+                                                                  arrayBufferView,
+                                                                  &isSharedMemory));
+    // The channel data arrays should all have originated in
+    // RestoreJSChannelData, where they are created unshared.
+    MOZ_ASSERT(!isSharedMemory);
+    auto stolenData = arrayBuffer
+      ? static_cast<float*>(JS_StealArrayBufferContents(aJSContext,
+                                                        arrayBuffer))
+      : nullptr;
+    if (stolenData) {
+      result->SetData(i, stolenData, js_free, stolenData);
+    } else {
+      NS_ASSERTION(i == 0, "some channels lost when contents not acquired");
+      return nullptr;
+    }
+  }
+
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    mJSChannels[i] = nullptr;
+  }
+
+  return result.forget();
+}
+
+ThreadSharedFloatArrayBufferList*
+AudioBuffer::GetThreadSharedChannelsForRate(JSContext* aJSContext)
+{
+  if (!mSharedChannels) {
+    mSharedChannels = StealJSArrayDataIntoSharedChannels(aJSContext);
+  }
+
+  return mSharedChannels;
+}
+
+size_t
+AudioBuffer::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = aMallocSizeOf(this);
+  amount += mJSChannels.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  if (mSharedChannels) {
+    amount += mSharedChannels->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioBuffer.h b/dom/media/webaudio/AudioBuffer.h
new file mode 100644
index 0000000000..2f2aef5fe8
--- /dev/null
+++ b/dom/media/webaudio/AudioBuffer.h
@@ -0,0 +1,137 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioBuffer_h_
+#define AudioBuffer_h_
+
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/StaticMutex.h"
+#include "nsTArray.h"
+#include "AudioContext.h"
+#include "js/TypeDecls.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+
+class ErrorResult;
+class ThreadSharedFloatArrayBufferList;
+
+namespace dom {
+
+class AudioContext;
+
+/**
+ * An AudioBuffer keeps its data either in the mJSChannels objects, which
+ * are Float32Arrays, or in mSharedChannels if the mJSChannels objects' buffers
+ * are detached.
+ */
+class AudioBuffer final : public nsWrapperCache
+{
+public:
+  // If non-null, aInitialContents must have number of channels equal to
+  // aNumberOfChannels and their lengths must be at least aLength.
+  static already_AddRefed<AudioBuffer>
+  Create(AudioContext* aContext, uint32_t aNumberOfChannels,
+         uint32_t aLength, float aSampleRate,
+         already_AddRefed<ThreadSharedFloatArrayBufferList> aInitialContents,
+         ErrorResult& aRv);
+
+  static already_AddRefed<AudioBuffer>
+  Create(AudioContext* aContext, uint32_t aNumberOfChannels,
+         uint32_t aLength, float aSampleRate,
+         ErrorResult& aRv)
+  {
+    return Create(aContext, aNumberOfChannels, aLength, aSampleRate,
+                  nullptr, aRv);
+  }
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioBuffer)
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(AudioBuffer)
+
+  nsPIDOMWindowInner* GetParentObject() const
+  {
+    nsCOMPtr<nsPIDOMWindowInner> parentObject = do_QueryReferent(mOwnerWindow);
+    return parentObject;
+  }
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  float SampleRate() const
+  {
+    return mSampleRate;
+  }
+
+  uint32_t Length() const
+  {
+    return mLength;
+  }
+
+  double Duration() const
+  {
+    return mLength / static_cast<double> (mSampleRate);
+  }
+
+  uint32_t NumberOfChannels() const
+  {
+    return mJSChannels.Length();
+  }
+
+  /**
+   * If mSharedChannels is non-null, copies its contents to
+   * new Float32Arrays in mJSChannels. Returns a Float32Array.
+   */
+  void GetChannelData(JSContext* aJSContext, uint32_t aChannel,
+                      JS::MutableHandle<JSObject*> aRetval,
+                      ErrorResult& aRv);
+
+  void CopyFromChannel(const Float32Array& aDestination, uint32_t aChannelNumber,
+                       uint32_t aStartInChannel, ErrorResult& aRv);
+  void CopyToChannel(JSContext* aJSContext, const Float32Array& aSource,
+                     uint32_t aChannelNumber, uint32_t aStartInChannel,
+                     ErrorResult& aRv);
+
+  /**
+   * Returns a ThreadSharedFloatArrayBufferList containing the sample data.
+   * Can return null if there is no data.
+   */
+  ThreadSharedFloatArrayBufferList* GetThreadSharedChannelsForRate(JSContext* aContext);
+
+protected:
+  AudioBuffer(AudioContext* aContext, uint32_t aNumberOfChannels,
+              uint32_t aLength, float aSampleRate,
+              already_AddRefed<ThreadSharedFloatArrayBufferList>
+                aInitialContents);
+  ~AudioBuffer();
+
+  bool RestoreJSChannelData(JSContext* aJSContext);
+
+  already_AddRefed<ThreadSharedFloatArrayBufferList>
+  StealJSArrayDataIntoSharedChannels(JSContext* aJSContext);
+
+  void ClearJSChannels();
+
+  nsWeakPtr mOwnerWindow;
+  // Float32Arrays
+  AutoTArray<JS::Heap<JSObject*>, 2> mJSChannels;
+
+  // mSharedChannels aggregates the data from mJSChannels. This is non-null
+  // if and only if the mJSChannels' buffers are detached.
+  RefPtr<ThreadSharedFloatArrayBufferList> mSharedChannels;
+
+  uint32_t mLength;
+  float mSampleRate;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioBufferSourceNode.cpp b/dom/media/webaudio/AudioBufferSourceNode.cpp
new file mode 100644
index 0000000000..51b6bab4a2
--- /dev/null
+++ b/dom/media/webaudio/AudioBufferSourceNode.cpp
@@ -0,0 +1,853 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioBufferSourceNode.h"
+#include "nsDebug.h"
+#include "mozilla/dom/AudioBufferSourceNodeBinding.h"
+#include "mozilla/dom/AudioParam.h"
+#include "mozilla/FloatingPoint.h"
+#include "nsContentUtils.h"
+#include "nsMathUtils.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "AudioParamTimeline.h"
+#include <limits>
+#include <algorithm>
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode, AudioNode, mBuffer, mPlaybackRate, mDetune)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
+
+/**
+ * Media-thread playback engine for AudioBufferSourceNode.
+ * Nothing is played until a non-null buffer has been set (via
+ * AudioNodeStream::SetBuffer) and a non-zero mBufferEnd has been set (via
+ * AudioNodeStream::SetInt32Parameter).
+ */
+class AudioBufferSourceNodeEngine final : public AudioNodeEngine
+{
+public:
+  AudioBufferSourceNodeEngine(AudioNode* aNode,
+                              AudioDestinationNode* aDestination) :
+    AudioNodeEngine(aNode),
+    mStart(0.0), mBeginProcessing(0),
+    mStop(STREAM_TIME_MAX),
+    mResampler(nullptr), mRemainingResamplerTail(0),
+    mBufferEnd(0),
+    mLoopStart(0), mLoopEnd(0),
+    mBufferPosition(0), mBufferSampleRate(0),
+    // mResamplerOutRate is initialized in UpdateResampler().
+    mChannels(0),
+    mDopplerShift(1.0f),
+    mDestination(aDestination->Stream()),
+    mPlaybackRateTimeline(1.0f),
+    mDetuneTimeline(0.0f),
+    mLoop(false)
+  {}
+
+  ~AudioBufferSourceNodeEngine()
+  {
+    if (mResampler) {
+      speex_resampler_destroy(mResampler);
+    }
+  }
+
+  void SetSourceStream(AudioNodeStream* aSource)
+  {
+    mSource = aSource;
+  }
+
+  void RecvTimelineEvent(uint32_t aIndex,
+                         dom::AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case AudioBufferSourceNode::PLAYBACKRATE:
+      mPlaybackRateTimeline.InsertEvent<int64_t>(aEvent);
+      break;
+    case AudioBufferSourceNode::DETUNE:
+      mDetuneTimeline.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
+    }
+  }
+  void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) override
+  {
+    switch (aIndex) {
+    case AudioBufferSourceNode::STOP: mStop = aParam; break;
+    default:
+      NS_ERROR("Bad AudioBufferSourceNodeEngine StreamTimeParameter");
+    }
+  }
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override
+  {
+    switch (aIndex) {
+    case AudioBufferSourceNode::START:
+      MOZ_ASSERT(!mStart, "Another START?");
+      mStart = aParam * mDestination->SampleRate();
+      // Round to nearest
+      mBeginProcessing = mStart + 0.5;
+      break;
+    case AudioBufferSourceNode::DOPPLERSHIFT:
+      mDopplerShift = (aParam <= 0 || mozilla::IsNaN(aParam)) ? 1.0 : aParam;
+      break;
+    default:
+      NS_ERROR("Bad AudioBufferSourceNodeEngine double parameter.");
+    };
+  }
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    switch (aIndex) {
+    case AudioBufferSourceNode::SAMPLE_RATE:
+      MOZ_ASSERT(aParam > 0);
+      mBufferSampleRate = aParam;
+      mSource->SetActive();
+      break;
+    case AudioBufferSourceNode::BUFFERSTART:
+      MOZ_ASSERT(aParam >= 0);
+      if (mBufferPosition == 0) {
+        mBufferPosition = aParam;
+      }
+      break;
+    case AudioBufferSourceNode::BUFFEREND:
+      MOZ_ASSERT(aParam >= 0);
+      mBufferEnd = aParam;
+      break;
+    case AudioBufferSourceNode::LOOP: mLoop = !!aParam; break;
+    case AudioBufferSourceNode::LOOPSTART:
+      MOZ_ASSERT(aParam >= 0);
+      mLoopStart = aParam;
+      break;
+    case AudioBufferSourceNode::LOOPEND:
+      MOZ_ASSERT(aParam >= 0);
+      mLoopEnd = aParam;
+      break;
+    default:
+      NS_ERROR("Bad AudioBufferSourceNodeEngine Int32Parameter");
+    }
+  }
+  void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer) override
+  {
+    mBuffer = aBuffer;
+  }
+
+  bool BegunResampling()
+  {
+    return mBeginProcessing == -STREAM_TIME_MAX;
+  }
+
+  void UpdateResampler(int32_t aOutRate, uint32_t aChannels)
+  {
+    if (mResampler &&
+        (aChannels != mChannels ||
+         // If the resampler has begun, then it will have moved
+         // mBufferPosition to after the samples it has read, but it hasn't
+         // output its buffered samples.  Keep using the resampler, even if
+         // the rates now match, so that this latent segment is output.
+         (aOutRate == mBufferSampleRate && !BegunResampling()))) {
+      speex_resampler_destroy(mResampler);
+      mResampler = nullptr;
+      mRemainingResamplerTail = 0;
+      mBeginProcessing = mStart + 0.5;
+    }
+
+    if (aChannels == 0 ||
+        (aOutRate == mBufferSampleRate && !mResampler)) {
+      mResamplerOutRate = aOutRate;
+      return;
+    }
+
+    if (!mResampler) {
+      mChannels = aChannels;
+      mResampler = speex_resampler_init(mChannels, mBufferSampleRate, aOutRate,
+                                        SPEEX_RESAMPLER_QUALITY_MIN,
+                                        nullptr);
+    } else {
+      if (mResamplerOutRate == aOutRate) {
+        return;
+      }
+      if (speex_resampler_set_rate(mResampler, mBufferSampleRate, aOutRate) != RESAMPLER_ERR_SUCCESS) {
+        NS_ASSERTION(false, "speex_resampler_set_rate failed");
+        return;
+      }
+    }
+
+    mResamplerOutRate = aOutRate;
+
+    if (!BegunResampling()) {
+      // Low pass filter effects from the resampler mean that samples before
+      // the start time are influenced by resampling the buffer.  The input
+      // latency indicates half the filter width.
+      int64_t inputLatency = speex_resampler_get_input_latency(mResampler);
+      uint32_t ratioNum, ratioDen;
+      speex_resampler_get_ratio(mResampler, &ratioNum, &ratioDen);
+      // The output subsample resolution supported in aligning the resampler
+      // is ratioNum.  First round the start time to the nearest subsample.
+      int64_t subsample = mStart * ratioNum + 0.5;
+      // Now include the leading effects of the filter, and round *up* to the
+      // next whole tick, because there is no effect on samples outside the
+      // filter width.
+      mBeginProcessing =
+        (subsample - inputLatency * ratioDen + ratioNum - 1) / ratioNum;
+    }
+  }
+
+  // Borrow a full buffer of size WEBAUDIO_BLOCK_SIZE from the source buffer
+  // at offset aSourceOffset.  This avoids copying memory.
+  void BorrowFromInputBuffer(AudioBlock* aOutput,
+                             uint32_t aChannels)
+  {
+    aOutput->SetBuffer(mBuffer);
+    aOutput->mChannelData.SetLength(aChannels);
+    for (uint32_t i = 0; i < aChannels; ++i) {
+      aOutput->mChannelData[i] = mBuffer->GetData(i) + mBufferPosition;
+    }
+    aOutput->mVolume = 1.0f;
+    aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
+  }
+
+  // Copy aNumberOfFrames frames from the source buffer at offset aSourceOffset
+  // and put it at offset aBufferOffset in the destination buffer.
+  void CopyFromInputBuffer(AudioBlock* aOutput,
+                           uint32_t aChannels,
+                           uintptr_t aOffsetWithinBlock,
+                           uint32_t aNumberOfFrames) {
+    for (uint32_t i = 0; i < aChannels; ++i) {
+      float* baseChannelData = aOutput->ChannelFloatsForWrite(i);
+      memcpy(baseChannelData + aOffsetWithinBlock,
+             mBuffer->GetData(i) + mBufferPosition,
+             aNumberOfFrames * sizeof(float));
+    }
+  }
+
+  // Resamples input data to an output buffer, according to |mBufferSampleRate| and
+  // the playbackRate/detune.
+  // The number of frames consumed/produced depends on the amount of space
+  // remaining in both the input and output buffer, and the playback rate (that
+  // is, the ratio between the output samplerate and the input samplerate).
+  void CopyFromInputBufferWithResampling(AudioBlock* aOutput,
+                                         uint32_t aChannels,
+                                         uint32_t* aOffsetWithinBlock,
+                                         uint32_t aAvailableInOutput,
+                                         StreamTime* aCurrentPosition,
+                                         uint32_t aBufferMax)
+  {
+    if (*aOffsetWithinBlock == 0) {
+      aOutput->AllocateChannels(aChannels);
+    }
+    SpeexResamplerState* resampler = mResampler;
+    MOZ_ASSERT(aChannels > 0);
+
+    if (mBufferPosition < aBufferMax) {
+      uint32_t availableInInputBuffer = aBufferMax - mBufferPosition;
+      uint32_t ratioNum, ratioDen;
+      speex_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
+      // Limit the number of input samples copied and possibly
+      // format-converted for resampling by estimating how many will be used.
+      // This may be a little small if still filling the resampler with
+      // initial data, but we'll get called again and it will work out.
+      uint32_t inputLimit = aAvailableInOutput * ratioNum / ratioDen + 10;
+      if (!BegunResampling()) {
+        // First time the resampler is used.
+        uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
+        inputLimit += inputLatency;
+        // If starting after mStart, then play from the beginning of the
+        // buffer, but correct for input latency.  If starting before mStart,
+        // then align the resampler so that the time corresponding to the
+        // first input sample is mStart.
+        int64_t skipFracNum = static_cast<int64_t>(inputLatency) * ratioDen;
+        double leadTicks = mStart - *aCurrentPosition;
+        if (leadTicks > 0.0) {
+          // Round to nearest output subsample supported by the resampler at
+          // these rates.
+          int64_t leadSubsamples = leadTicks * ratioNum + 0.5;
+          MOZ_ASSERT(leadSubsamples <= skipFracNum,
+                     "mBeginProcessing is wrong?");
+          skipFracNum -= leadSubsamples;
+        }
+        speex_resampler_set_skip_frac_num(resampler,
+                                  std::min<int64_t>(skipFracNum, UINT32_MAX));
+
+        mBeginProcessing = -STREAM_TIME_MAX;
+      }
+      inputLimit = std::min(inputLimit, availableInInputBuffer);
+
+      for (uint32_t i = 0; true; ) {
+        uint32_t inSamples = inputLimit;
+        const float* inputData = mBuffer->GetData(i) + mBufferPosition;
+
+        uint32_t outSamples = aAvailableInOutput;
+        float* outputData =
+          aOutput->ChannelFloatsForWrite(i) + *aOffsetWithinBlock;
+
+        WebAudioUtils::SpeexResamplerProcess(resampler, i,
+                                             inputData, &inSamples,
+                                             outputData, &outSamples);
+        if (++i == aChannels) {
+          mBufferPosition += inSamples;
+          MOZ_ASSERT(mBufferPosition <= mBufferEnd || mLoop);
+          *aOffsetWithinBlock += outSamples;
+          *aCurrentPosition += outSamples;
+          if (inSamples == availableInInputBuffer && !mLoop) {
+            // We'll feed in enough zeros to empty out the resampler's memory.
+            // This handles the output latency as well as capturing the low
+            // pass effects of the resample filter.
+            mRemainingResamplerTail =
+              2 * speex_resampler_get_input_latency(resampler) - 1;
+          }
+          return;
+        }
+      }
+    } else {
+      for (uint32_t i = 0; true; ) {
+        uint32_t inSamples = mRemainingResamplerTail;
+        uint32_t outSamples = aAvailableInOutput;
+        float* outputData =
+          aOutput->ChannelFloatsForWrite(i) + *aOffsetWithinBlock;
+
+        // AudioDataValue* for aIn selects the function that does not try to
+        // copy and format-convert input data.
+        WebAudioUtils::SpeexResamplerProcess(resampler, i,
+                         static_cast<AudioDataValue*>(nullptr), &inSamples,
+                         outputData, &outSamples);
+        if (++i == aChannels) {
+          MOZ_ASSERT(inSamples <= mRemainingResamplerTail);
+          mRemainingResamplerTail -= inSamples;
+          *aOffsetWithinBlock += outSamples;
+          *aCurrentPosition += outSamples;
+          break;
+        }
+      }
+    }
+  }
+
+  /**
+   * Fill aOutput with as many zero frames as we can, and advance
+   * aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
+   * This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
+   * aCurrentPosition past aMaxPos.  This function knows when it needs to
+   * allocate the output buffer, and also optimizes the case where it can avoid
+   * memory allocations.
+   */
+  void FillWithZeroes(AudioBlock* aOutput,
+                      uint32_t aChannels,
+                      uint32_t* aOffsetWithinBlock,
+                      StreamTime* aCurrentPosition,
+                      StreamTime aMaxPos)
+  {
+    MOZ_ASSERT(*aCurrentPosition < aMaxPos);
+    uint32_t numFrames =
+      std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
+                           aMaxPos - *aCurrentPosition);
+    if (numFrames == WEBAUDIO_BLOCK_SIZE || !aChannels) {
+      aOutput->SetNull(numFrames);
+    } else {
+      if (*aOffsetWithinBlock == 0) {
+        aOutput->AllocateChannels(aChannels);
+      }
+      WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
+    }
+    *aOffsetWithinBlock += numFrames;
+    *aCurrentPosition += numFrames;
+  }
+
+  /**
+   * Copy as many frames as possible from the source buffer to aOutput, and
+   * advance aOffsetWithinBlock and aCurrentPosition based on how many frames
+   * we write.  This will never advance aOffsetWithinBlock past
+   * WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop.  It takes data from
+   * the buffer at aBufferOffset, and never takes more data than aBufferMax.
+   * This function knows when it needs to allocate the output buffer, and also
+   * optimizes the case where it can avoid memory allocations.
+   */
+  void CopyFromBuffer(AudioBlock* aOutput,
+                      uint32_t aChannels,
+                      uint32_t* aOffsetWithinBlock,
+                      StreamTime* aCurrentPosition,
+                      uint32_t aBufferMax)
+  {
+    MOZ_ASSERT(*aCurrentPosition < mStop);
+    uint32_t availableInOutput =
+      std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
+                           mStop - *aCurrentPosition);
+    if (mResampler) {
+      CopyFromInputBufferWithResampling(aOutput, aChannels,
+                                        aOffsetWithinBlock, availableInOutput,
+                                        aCurrentPosition, aBufferMax);
+      return;
+    }
+
+    if (aChannels == 0) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      // There is no attempt here to limit advance so that mBufferPosition is
+      // limited to aBufferMax.  The only observable affect of skipping the
+      // check would be in the precise timing of the ended event if the loop
+      // attribute is reset after playback has looped.
+      *aOffsetWithinBlock += availableInOutput;
+      *aCurrentPosition += availableInOutput;
+      // Rounding at the start and end of the period means that fractional
+      // increments essentially accumulate if outRate remains constant.  If
+      // outRate is varying, then accumulation happens on average but not
+      // precisely.
+      TrackTicks start = *aCurrentPosition *
+        mBufferSampleRate / mResamplerOutRate;
+      TrackTicks end = (*aCurrentPosition + availableInOutput) *
+        mBufferSampleRate / mResamplerOutRate;
+      mBufferPosition += end - start;
+      return;
+    }
+
+    uint32_t numFrames = std::min(aBufferMax - mBufferPosition,
+                                  availableInOutput);
+
+    bool inputBufferAligned = true;
+    for (uint32_t i = 0; i < aChannels; ++i) {
+      if (!IS_ALIGNED16(mBuffer->GetData(i) + mBufferPosition)) {
+        inputBufferAligned = false;
+      }
+    }
+
+    if (numFrames == WEBAUDIO_BLOCK_SIZE && inputBufferAligned) {
+      MOZ_ASSERT(mBufferPosition < aBufferMax);
+      BorrowFromInputBuffer(aOutput, aChannels);
+    } else {
+      if (*aOffsetWithinBlock == 0) {
+        aOutput->AllocateChannels(aChannels);
+      }
+      MOZ_ASSERT(mBufferPosition < aBufferMax);
+      CopyFromInputBuffer(aOutput, aChannels, *aOffsetWithinBlock, numFrames);
+    }
+    *aOffsetWithinBlock += numFrames;
+    *aCurrentPosition += numFrames;
+    mBufferPosition += numFrames;
+  }
+
+  int32_t ComputeFinalOutSampleRate(float aPlaybackRate, float aDetune)
+  {
+    float computedPlaybackRate = aPlaybackRate * pow(2, aDetune / 1200.f);
+    // Make sure the playback rate and the doppler shift are something
+    // our resampler can work with.
+    int32_t rate = WebAudioUtils::
+      TruncateFloatToInt<int32_t>(mSource->SampleRate() /
+                                  (computedPlaybackRate * mDopplerShift));
+    return rate ? rate : mBufferSampleRate;
+  }
+
+  void UpdateSampleRateIfNeeded(uint32_t aChannels, StreamTime aStreamPosition)
+  {
+    float playbackRate;
+    float detune;
+
+    if (mPlaybackRateTimeline.HasSimpleValue()) {
+      playbackRate = mPlaybackRateTimeline.GetValue();
+    } else {
+      playbackRate = mPlaybackRateTimeline.GetValueAtTime(aStreamPosition);
+    }
+    if (mDetuneTimeline.HasSimpleValue()) {
+      detune = mDetuneTimeline.GetValue();
+    } else {
+      detune = mDetuneTimeline.GetValueAtTime(aStreamPosition);
+    }
+    if (playbackRate <= 0 || mozilla::IsNaN(playbackRate)) {
+      playbackRate = 1.0f;
+    }
+
+    detune = std::min(std::max(-1200.f, detune), 1200.f);
+
+    int32_t outRate = ComputeFinalOutSampleRate(playbackRate, detune);
+    UpdateResampler(outRate, aChannels);
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    if (mBufferSampleRate == 0) {
+      // start() has not yet been called or no buffer has yet been set
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    StreamTime streamPosition = mDestination->GraphTimeToStreamTime(aFrom);
+    uint32_t channels = mBuffer ? mBuffer->GetChannels() : 0;
+
+    UpdateSampleRateIfNeeded(channels, streamPosition);
+
+    uint32_t written = 0;
+    while (written < WEBAUDIO_BLOCK_SIZE) {
+      if (mStop != STREAM_TIME_MAX &&
+          streamPosition >= mStop) {
+        FillWithZeroes(aOutput, channels, &written, &streamPosition, STREAM_TIME_MAX);
+        continue;
+      }
+      if (streamPosition < mBeginProcessing) {
+        FillWithZeroes(aOutput, channels, &written, &streamPosition,
+                       mBeginProcessing);
+        continue;
+      }
+      if (mLoop) {
+        // mLoopEnd can become less than mBufferPosition when a LOOPEND engine
+        // parameter is received after "loopend" is changed on the node or a
+        // new buffer with lower samplerate is set.
+        if (mBufferPosition >= mLoopEnd) {
+          mBufferPosition = mLoopStart;
+        }
+        CopyFromBuffer(aOutput, channels, &written, &streamPosition, mLoopEnd);
+      } else {
+        if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
+          CopyFromBuffer(aOutput, channels, &written, &streamPosition, mBufferEnd);
+        } else {
+          FillWithZeroes(aOutput, channels, &written, &streamPosition, STREAM_TIME_MAX);
+        }
+      }
+    }
+
+    // We've finished if we've gone past mStop, or if we're past mDuration when
+    // looping is disabled.
+    if (streamPosition >= mStop ||
+        (!mLoop && mBufferPosition >= mBufferEnd && !mRemainingResamplerTail)) {
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // Whether buffer has been set and start() has been called.
+    return mBufferSampleRate != 0;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mBuffer - shared w/ AudioNode
+    // - mPlaybackRateTimeline - shared w/ AudioNode
+    // - mDetuneTimeline - shared w/ AudioNode
+
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    // NB: We need to modify speex if we want the full memory picture, internal
+    //     fields that need measuring noted below.
+    // - mResampler->mem
+    // - mResampler->sinc_table
+    // - mResampler->last_sample
+    // - mResampler->magic_samples
+    // - mResampler->samp_frac_num
+    amount += aMallocSizeOf(mResampler);
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  double mStart; // including the fractional position between ticks
+  // Low pass filter effects from the resampler mean that samples before the
+  // start time are influenced by resampling the buffer.  mBeginProcessing
+  // includes the extent of this filter.  The special value of -STREAM_TIME_MAX
+  // indicates that the resampler has begun processing.
+  StreamTime mBeginProcessing;
+  StreamTime mStop;
+  RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  SpeexResamplerState* mResampler;
+  // mRemainingResamplerTail, like mBufferPosition, and
+  // mBufferEnd, is measured in input buffer samples.
+  uint32_t mRemainingResamplerTail;
+  uint32_t mBufferEnd;
+  uint32_t mLoopStart;
+  uint32_t mLoopEnd;
+  uint32_t mBufferPosition;
+  int32_t mBufferSampleRate;
+  int32_t mResamplerOutRate;
+  uint32_t mChannels;
+  float mDopplerShift;
+  AudioNodeStream* mDestination;
+  AudioNodeStream* mSource;
+  AudioParamTimeline mPlaybackRateTimeline;
+  AudioParamTimeline mDetuneTimeline;
+  bool mLoop;
+};
+
+AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mLoopStart(0.0)
+  , mLoopEnd(0.0)
+  // mOffset and mDuration are initialized in Start().
+  , mPlaybackRate(new AudioParam(this, PLAYBACKRATE, 1.0f, "playbackRate"))
+  , mDetune(new AudioParam(this, DETUNE, 0.0f, "detune"))
+  , mLoop(false)
+  , mStartCalled(false)
+{
+  AudioBufferSourceNodeEngine* engine = new AudioBufferSourceNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NEED_MAIN_THREAD_FINISHED,
+                                    aContext->Graph());
+  engine->SetSourceStream(mStream);
+  mStream->AddMainThreadListener(this);
+}
+
+AudioBufferSourceNode::~AudioBufferSourceNode()
+{
+}
+
+void
+AudioBufferSourceNode::DestroyMediaStream()
+{
+  bool hadStream = mStream;
+  if (hadStream) {
+    mStream->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaStream();
+  if (hadStream && Context()) {
+    Context()->UnregisterAudioBufferSourceNode(this);
+  }
+}
+
+size_t
+AudioBufferSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  /* mBuffer can be shared and is accounted for separately. */
+
+  amount += mPlaybackRate->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+AudioBufferSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+AudioBufferSourceNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioBufferSourceNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+AudioBufferSourceNode::Start(double aWhen, double aOffset,
+                             const Optional<double>& aDuration, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen) ||
+      (aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value()))) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+  mStartCalled = true;
+
+  AudioNodeStream* ns = mStream;
+  if (!ns) {
+    // Nothing to play, or we're already dead for some reason
+    return;
+  }
+
+  // Remember our arguments so that we can use them when we get a new buffer.
+  mOffset = aOffset;
+  mDuration = aDuration.WasPassed() ? aDuration.Value()
+                                    : std::numeric_limits<double>::min();
+
+  WEB_AUDIO_API_LOG("%f: %s %u Start(%f, %g, %g)", Context()->CurrentTime(),
+                    NodeType(), Id(), aWhen, aOffset, mDuration);
+
+  // We can't send these parameters without a buffer because we don't know the
+  // buffer's sample rate or length.
+  if (mBuffer) {
+    SendOffsetAndDurationParametersToStream(ns);
+  }
+
+  // Don't set parameter unnecessarily
+  if (aWhen > 0.0) {
+    ns->SetDoubleParameter(START, aWhen);
+  }
+}
+
+void
+AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
+{
+  AudioNodeStream* ns = mStream;
+  if (!ns) {
+    return;
+  }
+
+  if (mBuffer) {
+    RefPtr<ThreadSharedFloatArrayBufferList> data =
+      mBuffer->GetThreadSharedChannelsForRate(aCx);
+    ns->SetBuffer(data.forget());
+
+    if (mStartCalled) {
+      SendOffsetAndDurationParametersToStream(ns);
+    }
+  } else {
+    ns->SetInt32Parameter(BUFFEREND, 0);
+    ns->SetBuffer(nullptr);
+
+    MarkInactive();
+  }
+}
+
+void
+AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
+{
+  NS_ASSERTION(mBuffer && mStartCalled,
+               "Only call this when we have a buffer and start() has been called");
+
+  float rate = mBuffer->SampleRate();
+  aStream->SetInt32Parameter(SAMPLE_RATE, rate);
+
+  int32_t bufferEnd = mBuffer->Length();
+  int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
+
+  // Don't set parameter unnecessarily
+  if (offsetSamples > 0) {
+    aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
+  }
+
+  if (mDuration != std::numeric_limits<double>::min()) {
+    MOZ_ASSERT(mDuration >= 0.0); // provided by Start()
+    MOZ_ASSERT(rate >= 0.0f); // provided by AudioBuffer::Create()
+    static_assert(std::numeric_limits<double>::digits >=
+                  std::numeric_limits<decltype(bufferEnd)>::digits,
+                  "bufferEnd should be represented exactly by double");
+    // + 0.5 rounds mDuration to nearest sample when assigned to bufferEnd.
+    bufferEnd = std::min<double>(bufferEnd,
+                                 offsetSamples + mDuration * rate + 0.5);
+  }
+  aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
+
+  MarkActive();
+}
+
+void
+AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+
+  WEB_AUDIO_API_LOG("%f: %s %u Stop(%f)", Context()->CurrentTime(),
+                    NodeType(), Id(), aWhen);
+
+  AudioNodeStream* ns = mStream;
+  if (!ns || !Context()) {
+    // We've already stopped and had our stream shut down
+    return;
+  }
+
+  ns->SetStreamTimeParameter(STOP, Context(), std::max(0.0, aWhen));
+}
+
+void
+AudioBufferSourceNode::NotifyMainThreadStreamFinished()
+{
+  MOZ_ASSERT(mStream->IsFinished());
+
+  class EndedEventDispatcher final : public Runnable
+  {
+  public:
+    explicit EndedEventDispatcher(AudioBufferSourceNode* aNode)
+      : mNode(aNode) {}
+    NS_IMETHOD Run() override
+    {
+      // If it's not safe to run scripts right now, schedule this to run later
+      if (!nsContentUtils::IsSafeToRunScript()) {
+        nsContentUtils::AddScriptRunner(this);
+        return NS_OK;
+      }
+
+      mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
+      // Release stream resources.
+      mNode->DestroyMediaStream();
+      return NS_OK;
+    }
+  private:
+    RefPtr<AudioBufferSourceNode> mNode;
+  };
+
+  NS_DispatchToMainThread(new EndedEventDispatcher(this));
+
+  // Drop the playing reference
+  // Warning: The below line might delete this.
+  MarkInactive();
+}
+
+void
+AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift)
+{
+  MOZ_ASSERT(mStream, "Should have disconnected panner if no stream");
+  SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
+}
+
+void
+AudioBufferSourceNode::SendLoopParametersToStream()
+{
+  if (!mStream) {
+    return;
+  }
+  // Don't compute and set the loop parameters unnecessarily
+  if (mLoop && mBuffer) {
+    float rate = mBuffer->SampleRate();
+    double length = (double(mBuffer->Length()) / mBuffer->SampleRate());
+    double actualLoopStart, actualLoopEnd;
+    if (mLoopStart >= 0.0 && mLoopEnd > 0.0 &&
+        mLoopStart < mLoopEnd) {
+      MOZ_ASSERT(mLoopStart != 0.0 || mLoopEnd != 0.0);
+      actualLoopStart = (mLoopStart > length) ? 0.0 : mLoopStart;
+      actualLoopEnd = std::min(mLoopEnd, length);
+    } else {
+      actualLoopStart = 0.0;
+      actualLoopEnd = length;
+    }
+    int32_t loopStartTicks = NS_lround(actualLoopStart * rate);
+    int32_t loopEndTicks = NS_lround(actualLoopEnd * rate);
+    if (loopStartTicks < loopEndTicks) {
+      SendInt32ParameterToStream(LOOPSTART, loopStartTicks);
+      SendInt32ParameterToStream(LOOPEND, loopEndTicks);
+      SendInt32ParameterToStream(LOOP, 1);
+    } else {
+      // Be explicit about looping not happening if the offsets make
+      // looping impossible.
+      SendInt32ParameterToStream(LOOP, 0);
+    }
+  } else {
+    SendInt32ParameterToStream(LOOP, 0);
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioBufferSourceNode.h b/dom/media/webaudio/AudioBufferSourceNode.h
new file mode 100644
index 0000000000..d982ec5cce
--- /dev/null
+++ b/dom/media/webaudio/AudioBufferSourceNode.h
@@ -0,0 +1,149 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioBufferSourceNode_h_
+#define AudioBufferSourceNode_h_
+
+#include "AudioNode.h"
+#include "AudioBuffer.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioParam;
+
+class AudioBufferSourceNode final : public AudioNode,
+                                    public MainThreadMediaStreamListener
+{
+public:
+  explicit AudioBufferSourceNode(AudioContext* aContext);
+
+  void DestroyMediaStream() override;
+
+  uint16_t NumberOfInputs() const final override
+  {
+    return 0;
+  }
+  AudioBufferSourceNode* AsAudioBufferSourceNode() override
+  {
+    return this;
+  }
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioBufferSourceNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void Start(double aWhen, double aOffset,
+             const Optional<double>& aDuration, ErrorResult& aRv);
+  void Stop(double aWhen, ErrorResult& aRv);
+
+  AudioBuffer* GetBuffer(JSContext* aCx) const
+  {
+    return mBuffer;
+  }
+  void SetBuffer(JSContext* aCx, AudioBuffer* aBuffer)
+  {
+    mBuffer = aBuffer;
+    SendBufferParameterToStream(aCx);
+    SendLoopParametersToStream();
+  }
+  AudioParam* PlaybackRate() const
+  {
+    return mPlaybackRate;
+  }
+  AudioParam* Detune() const
+  {
+    return mDetune;
+  }
+  bool Loop() const
+  {
+    return mLoop;
+  }
+  void SetLoop(bool aLoop)
+  {
+    mLoop = aLoop;
+    SendLoopParametersToStream();
+  }
+  double LoopStart() const
+  {
+    return mLoopStart;
+  }
+  void SetLoopStart(double aStart)
+  {
+    mLoopStart = aStart;
+    SendLoopParametersToStream();
+  }
+  double LoopEnd() const
+  {
+    return mLoopEnd;
+  }
+  void SetLoopEnd(double aEnd)
+  {
+    mLoopEnd = aEnd;
+    SendLoopParametersToStream();
+  }
+  void SendDopplerShiftToStream(double aDopplerShift);
+
+  IMPL_EVENT_HANDLER(ended)
+
+  void NotifyMainThreadStreamFinished() override;
+
+  const char* NodeType() const override
+  {
+    return "AudioBufferSourceNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~AudioBufferSourceNode();
+
+private:
+  friend class AudioBufferSourceNodeEngine;
+  // START is sent during Start().
+  // STOP is sent during Stop().
+  // BUFFERSTART and BUFFEREND are sent when SetBuffer() and Start() have
+  // been called (along with sending the buffer).
+  enum EngineParameters {
+    SAMPLE_RATE,
+    START,
+    STOP,
+    // BUFFERSTART is the "offset" passed to start(), multiplied by
+    // buffer.sampleRate.
+    BUFFERSTART,
+    // BUFFEREND is the sum of "offset" and "duration" passed to start(),
+    // multiplied by buffer.sampleRate, or the size of the buffer, if smaller.
+    BUFFEREND,
+    LOOP,
+    LOOPSTART,
+    LOOPEND,
+    PLAYBACKRATE,
+    DETUNE,
+    DOPPLERSHIFT
+  };
+
+  void SendLoopParametersToStream();
+  void SendBufferParameterToStream(JSContext* aCx);
+  void SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream);
+
+private:
+  double mLoopStart;
+  double mLoopEnd;
+  double mOffset;
+  double mDuration;
+  RefPtr<AudioBuffer> mBuffer;
+  RefPtr<AudioParam> mPlaybackRate;
+  RefPtr<AudioParam> mDetune;
+  bool mLoop;
+  bool mStartCalled;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioContext.cpp b/dom/media/webaudio/AudioContext.cpp
new file mode 100644
index 0000000000..f61226a482
--- /dev/null
+++ b/dom/media/webaudio/AudioContext.cpp
@@ -0,0 +1,1247 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioContext.h"
+
+#include "blink/PeriodicWave.h"
+
+#include "mozilla/ErrorResult.h"
+#include "mozilla/OwningNonNull.h"
+
+#include "mozilla/dom/AnalyserNode.h"
+#include "mozilla/dom/AudioContextBinding.h"
+#include "mozilla/dom/HTMLMediaElement.h"
+#include "mozilla/dom/OfflineAudioContextBinding.h"
+#include "mozilla/dom/Promise.h"
+
+#include "AudioBuffer.h"
+#include "AudioBufferSourceNode.h"
+#include "AudioChannelService.h"
+#include "AudioDestinationNode.h"
+#include "AudioListener.h"
+#include "AudioStream.h"
+#include "BiquadFilterNode.h"
+#include "ChannelMergerNode.h"
+#include "ChannelSplitterNode.h"
+#include "ConstantSourceNode.h"
+#include "ConvolverNode.h"
+#include "DelayNode.h"
+#include "DynamicsCompressorNode.h"
+#include "GainNode.h"
+#include "IIRFilterNode.h"
+#include "MediaElementAudioSourceNode.h"
+#include "MediaStreamAudioDestinationNode.h"
+#include "MediaStreamAudioSourceNode.h"
+#include "MediaStreamGraph.h"
+#include "nsContentUtils.h"
+#include "nsNetCID.h"
+#include "nsNetUtil.h"
+#include "nsPIDOMWindow.h"
+#include "nsPrintfCString.h"
+#include "OscillatorNode.h"
+#include "PannerNode.h"
+#include "PeriodicWave.h"
+#include "ScriptProcessorNode.h"
+#include "StereoPannerNode.h"
+#include "WaveShaperNode.h"
+
+namespace mozilla {
+namespace dom {
+
+// 0 is a special value that MediaStreams use to denote they are not part of a
+// AudioContext.
+static dom::AudioContext::AudioContextId gAudioContextId = 1;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioContext)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioContext)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mDestination)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mListener)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mPromiseGripArray)
+  if (!tmp->mIsStarted) {
+    NS_IMPL_CYCLE_COLLECTION_UNLINK(mActiveNodes)
+  }
+  // mDecodeJobs owns the WebAudioDecodeJob objects whose lifetime is managed explicitly.
+  // mAllNodes is an array of weak pointers, ignore it here.
+  // mPannerNodes is an array of weak pointers, ignore it here.
+  // mBasicWaveFormCache cannot participate in cycles, ignore it here.
+
+  // Remove weak reference on the global window as the context is not usable
+  // without mDestination.
+  tmp->DisconnectFromWindow();
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(DOMEventTargetHelper)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioContext,
+                                                  DOMEventTargetHelper)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mDestination)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mListener)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPromiseGripArray)
+  if (!tmp->mIsStarted) {
+    MOZ_ASSERT(tmp->mIsOffline,
+               "Online AudioContexts should always be started");
+    NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mActiveNodes)
+  }
+  // mDecodeJobs owns the WebAudioDecodeJob objects whose lifetime is managed explicitly.
+  // mAllNodes is an array of weak pointers, ignore it here.
+  // mPannerNodes is an array of weak pointers, ignore it here.
+  // mBasicWaveFormCache cannot participate in cycles, ignore it here.
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_ADDREF_INHERITED(AudioContext, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(AudioContext, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioContext)
+  NS_INTERFACE_MAP_ENTRY(nsIMemoryReporter)
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+static float GetSampleRateForAudioContext(bool aIsOffline, float aSampleRate)
+{
+  if (aIsOffline) {
+    return aSampleRate;
+  } else {
+    return static_cast<float>(CubebUtils::PreferredSampleRate());
+  }
+}
+
+AudioContext::AudioContext(nsPIDOMWindowInner* aWindow,
+                           bool aIsOffline,
+                           AudioChannel aChannel,
+                           uint32_t aNumberOfChannels,
+                           uint32_t aLength,
+                           float aSampleRate)
+  : DOMEventTargetHelper(aWindow)
+  , mId(gAudioContextId++)
+  , mSampleRate(GetSampleRateForAudioContext(aIsOffline, aSampleRate))
+  , mAudioContextState(AudioContextState::Suspended)
+  , mNumberOfChannels(aNumberOfChannels)
+  , mIsOffline(aIsOffline)
+  , mIsStarted(!aIsOffline)
+  , mIsShutDown(false)
+  , mCloseCalled(false)
+  , mSuspendCalled(false)
+{
+  bool mute = aWindow->AddAudioContext(this);
+
+  // Note: AudioDestinationNode needs an AudioContext that must already be
+  // bound to the window.
+  mDestination = new AudioDestinationNode(this, aIsOffline, aChannel,
+                                          aNumberOfChannels, aLength, aSampleRate);
+
+  // The context can't be muted until it has a destination.
+  if (mute) {
+    Mute();
+  }
+}
+
+nsresult
+AudioContext::Init()
+{
+  if (!mIsOffline) {
+    nsresult rv = mDestination->CreateAudioChannelAgent();
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      return rv;
+    }
+  }
+
+  return NS_OK;
+}
+
+void
+AudioContext::DisconnectFromWindow()
+{
+  nsPIDOMWindowInner* window = GetOwner();
+  if (window) {
+    window->RemoveAudioContext(this);
+  }
+}
+
+AudioContext::~AudioContext()
+{
+  DisconnectFromWindow();
+  UnregisterWeakMemoryReporter(this);
+}
+
+JSObject*
+AudioContext::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  if (mIsOffline) {
+    return OfflineAudioContextBinding::Wrap(aCx, this, aGivenProto);
+  } else {
+    return AudioContextBinding::Wrap(aCx, this, aGivenProto);
+  }
+}
+
+/* static */ already_AddRefed<AudioContext>
+AudioContext::Constructor(const GlobalObject& aGlobal,
+                          ErrorResult& aRv)
+{
+  return AudioContext::Constructor(aGlobal,
+                                   AudioChannelService::GetDefaultAudioChannel(),
+                                   aRv);
+}
+
+/* static */ already_AddRefed<AudioContext>
+AudioContext::Constructor(const GlobalObject& aGlobal,
+                          AudioChannel aChannel,
+                          ErrorResult& aRv)
+{
+  nsCOMPtr<nsPIDOMWindowInner> window = do_QueryInterface(aGlobal.GetAsSupports());
+  if (!window) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+
+  RefPtr<AudioContext> object = new AudioContext(window, false, aChannel);
+  aRv = object->Init();
+  if (NS_WARN_IF(aRv.Failed())) {
+     return nullptr;
+  }
+
+  RegisterWeakMemoryReporter(object);
+
+  return object.forget();
+}
+
+/* static */ already_AddRefed<AudioContext>
+AudioContext::Constructor(const GlobalObject& aGlobal,
+                          uint32_t aNumberOfChannels,
+                          uint32_t aLength,
+                          float aSampleRate,
+                          ErrorResult& aRv)
+{
+  nsCOMPtr<nsPIDOMWindowInner> window = do_QueryInterface(aGlobal.GetAsSupports());
+  if (!window) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+
+  if (aNumberOfChannels == 0 ||
+      aNumberOfChannels > WebAudioUtils::MaxChannelCount ||
+      aLength == 0 ||
+      aSampleRate < WebAudioUtils::MinSampleRate ||
+      aSampleRate > WebAudioUtils::MaxSampleRate) {
+    // The DOM binding protects us against infinity and NaN
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  RefPtr<AudioContext> object = new AudioContext(window,
+                                                   true,
+                                                   AudioChannel::Normal,
+                                                   aNumberOfChannels,
+                                                   aLength,
+                                                   aSampleRate);
+
+  RegisterWeakMemoryReporter(object);
+
+  return object.forget();
+}
+
+bool AudioContext::CheckClosed(ErrorResult& aRv)
+{
+  if (mAudioContextState == AudioContextState::Closed) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return true;
+  }
+  return false;
+}
+
+already_AddRefed<AudioBufferSourceNode>
+AudioContext::CreateBufferSource(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<AudioBufferSourceNode> bufferNode =
+    new AudioBufferSourceNode(this);
+  return bufferNode.forget();
+}
+
+already_AddRefed<ConstantSourceNode>
+AudioContext::CreateConstantSource(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<ConstantSourceNode> constantSourceNode =
+    new ConstantSourceNode(this);
+  return constantSourceNode.forget();
+}
+
+already_AddRefed<AudioBuffer>
+AudioContext::CreateBuffer(uint32_t aNumberOfChannels, uint32_t aLength,
+                           float aSampleRate,
+                           ErrorResult& aRv)
+{
+  if (!aNumberOfChannels) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  return AudioBuffer::Create(this, aNumberOfChannels, aLength,
+                             aSampleRate, aRv);
+}
+
+namespace {
+
+bool IsValidBufferSize(uint32_t aBufferSize) {
+  switch (aBufferSize) {
+  case 0:       // let the implementation choose the buffer size
+  case 256:
+  case 512:
+  case 1024:
+  case 2048:
+  case 4096:
+  case 8192:
+  case 16384:
+    return true;
+  default:
+    return false;
+  }
+}
+
+} // namespace
+
+already_AddRefed<MediaStreamAudioDestinationNode>
+AudioContext::CreateMediaStreamDestination(ErrorResult& aRv)
+{
+  if (mIsOffline) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<MediaStreamAudioDestinationNode> node =
+      new MediaStreamAudioDestinationNode(this);
+  return node.forget();
+}
+
+already_AddRefed<ScriptProcessorNode>
+AudioContext::CreateScriptProcessor(uint32_t aBufferSize,
+                                    uint32_t aNumberOfInputChannels,
+                                    uint32_t aNumberOfOutputChannels,
+                                    ErrorResult& aRv)
+{
+  if ((aNumberOfInputChannels == 0 && aNumberOfOutputChannels == 0) ||
+      aNumberOfInputChannels > WebAudioUtils::MaxChannelCount ||
+      aNumberOfOutputChannels > WebAudioUtils::MaxChannelCount ||
+      !IsValidBufferSize(aBufferSize)) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<ScriptProcessorNode> scriptProcessor =
+    new ScriptProcessorNode(this, aBufferSize, aNumberOfInputChannels,
+                            aNumberOfOutputChannels);
+  return scriptProcessor.forget();
+}
+
+already_AddRefed<AnalyserNode>
+AudioContext::CreateAnalyser(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<AnalyserNode> analyserNode = new AnalyserNode(this);
+  return analyserNode.forget();
+}
+
+already_AddRefed<StereoPannerNode>
+AudioContext::CreateStereoPanner(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<StereoPannerNode> stereoPannerNode = new StereoPannerNode(this);
+  return stereoPannerNode.forget();
+}
+
+already_AddRefed<MediaElementAudioSourceNode>
+AudioContext::CreateMediaElementSource(HTMLMediaElement& aMediaElement,
+                                       ErrorResult& aRv)
+{
+  if (mIsOffline) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  if (aMediaElement.ContainsRestrictedContent()) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<DOMMediaStream> stream =
+    aMediaElement.CaptureAudio(aRv, mDestination->Stream()->Graph());
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  return MediaElementAudioSourceNode::Create(this, stream, aRv);
+}
+
+already_AddRefed<MediaStreamAudioSourceNode>
+AudioContext::CreateMediaStreamSource(DOMMediaStream& aMediaStream,
+                                      ErrorResult& aRv)
+{
+  if (mIsOffline) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  return MediaStreamAudioSourceNode::Create(this, &aMediaStream, aRv);
+}
+
+already_AddRefed<GainNode>
+AudioContext::CreateGain(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<GainNode> gainNode = new GainNode(this);
+  return gainNode.forget();
+}
+
+already_AddRefed<WaveShaperNode>
+AudioContext::CreateWaveShaper(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<WaveShaperNode> waveShaperNode = new WaveShaperNode(this);
+  return waveShaperNode.forget();
+}
+
+already_AddRefed<DelayNode>
+AudioContext::CreateDelay(double aMaxDelayTime, ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  if (aMaxDelayTime > 0. && aMaxDelayTime < 180.) {
+    RefPtr<DelayNode> delayNode = new DelayNode(this, aMaxDelayTime);
+    return delayNode.forget();
+  }
+
+  aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+  return nullptr;
+}
+
+already_AddRefed<PannerNode>
+AudioContext::CreatePanner(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<PannerNode> pannerNode = new PannerNode(this);
+  mPannerNodes.PutEntry(pannerNode);
+  return pannerNode.forget();
+}
+
+already_AddRefed<ConvolverNode>
+AudioContext::CreateConvolver(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<ConvolverNode> convolverNode = new ConvolverNode(this);
+  return convolverNode.forget();
+}
+
+already_AddRefed<ChannelSplitterNode>
+AudioContext::CreateChannelSplitter(uint32_t aNumberOfOutputs, ErrorResult& aRv)
+{
+  if (aNumberOfOutputs == 0 ||
+      aNumberOfOutputs > WebAudioUtils::MaxChannelCount) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<ChannelSplitterNode> splitterNode =
+    new ChannelSplitterNode(this, aNumberOfOutputs);
+  return splitterNode.forget();
+}
+
+already_AddRefed<ChannelMergerNode>
+AudioContext::CreateChannelMerger(uint32_t aNumberOfInputs, ErrorResult& aRv)
+{
+  if (aNumberOfInputs == 0 ||
+      aNumberOfInputs > WebAudioUtils::MaxChannelCount) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<ChannelMergerNode> mergerNode =
+    new ChannelMergerNode(this, aNumberOfInputs);
+  return mergerNode.forget();
+}
+
+already_AddRefed<DynamicsCompressorNode>
+AudioContext::CreateDynamicsCompressor(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<DynamicsCompressorNode> compressorNode =
+    new DynamicsCompressorNode(this);
+  return compressorNode.forget();
+}
+
+already_AddRefed<BiquadFilterNode>
+AudioContext::CreateBiquadFilter(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<BiquadFilterNode> filterNode =
+    new BiquadFilterNode(this);
+  return filterNode.forget();
+}
+
+already_AddRefed<IIRFilterNode>
+AudioContext::CreateIIRFilter(const mozilla::dom::binding_detail::AutoSequence<double>& aFeedforward,
+                              const mozilla::dom::binding_detail::AutoSequence<double>& aFeedback,
+                              mozilla::ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  if (aFeedforward.Length() == 0 || aFeedforward.Length() > 20) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  if (aFeedback.Length() == 0 || aFeedback.Length() > 20) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  bool feedforwardAllZeros = true;
+  for (size_t i = 0; i < aFeedforward.Length(); ++i) {
+    if (aFeedforward.Elements()[i] != 0.0) {
+      feedforwardAllZeros = false;
+    }
+  }
+
+  if (feedforwardAllZeros || aFeedback.Elements()[0] == 0.0) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return nullptr;
+  }
+
+  RefPtr<IIRFilterNode> filterNode =
+    new IIRFilterNode(this, aFeedforward, aFeedback);
+  return filterNode.forget();
+}
+
+already_AddRefed<OscillatorNode>
+AudioContext::CreateOscillator(ErrorResult& aRv)
+{
+  if (CheckClosed(aRv)) {
+    return nullptr;
+  }
+
+  RefPtr<OscillatorNode> oscillatorNode =
+    new OscillatorNode(this);
+  return oscillatorNode.forget();
+}
+
+already_AddRefed<PeriodicWave>
+AudioContext::CreatePeriodicWave(const Float32Array& aRealData,
+                                 const Float32Array& aImagData,
+                                 const PeriodicWaveConstraints& aConstraints,
+                                 ErrorResult& aRv)
+{
+  aRealData.ComputeLengthAndData();
+  aImagData.ComputeLengthAndData();
+
+  if (aRealData.Length() != aImagData.Length() ||
+      aRealData.Length() == 0) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return nullptr;
+  }
+
+  RefPtr<PeriodicWave> periodicWave =
+    new PeriodicWave(this, aRealData.Data(), aImagData.Data(),
+                     aImagData.Length(), aConstraints.mDisableNormalization,
+                     aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  return periodicWave.forget();
+}
+
+AudioListener*
+AudioContext::Listener()
+{
+  if (!mListener) {
+    mListener = new AudioListener(this);
+  }
+  return mListener;
+}
+
+already_AddRefed<Promise>
+AudioContext::DecodeAudioData(const ArrayBuffer& aBuffer,
+                              const Optional<OwningNonNull<DecodeSuccessCallback> >& aSuccessCallback,
+                              const Optional<OwningNonNull<DecodeErrorCallback> >& aFailureCallback,
+                              ErrorResult& aRv)
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+  RefPtr<Promise> promise;
+  AutoJSAPI jsapi;
+  jsapi.Init();
+  JSContext* cx = jsapi.cx();
+  JSAutoCompartment ac(cx, aBuffer.Obj());
+
+  promise = Promise::Create(parentObject, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  aBuffer.ComputeLengthAndData();
+
+  if (aBuffer.IsShared()) {
+    // Throw if the object is mapping shared memory (must opt in).
+    aRv.ThrowTypeError<MSG_TYPEDARRAY_IS_SHARED>(NS_LITERAL_STRING("Argument of AudioContext.decodeAudioData"));
+    return nullptr;
+  }
+
+  // Detach the array buffer
+  size_t length = aBuffer.Length();
+  JS::RootedObject obj(cx, aBuffer.Obj());
+
+  uint8_t* data = static_cast<uint8_t*>(JS_StealArrayBufferContents(cx, obj));
+
+  // Sniff the content of the media.
+  // Failed type sniffing will be handled by AsyncDecodeWebAudio.
+  nsAutoCString contentType;
+  NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr, data, length, contentType);
+
+  RefPtr<DecodeErrorCallback> failureCallback;
+  RefPtr<DecodeSuccessCallback> successCallback;
+  if (aFailureCallback.WasPassed()) {
+    failureCallback = &aFailureCallback.Value();
+  }
+  if (aSuccessCallback.WasPassed()) {
+    successCallback = &aSuccessCallback.Value();
+  }
+  RefPtr<WebAudioDecodeJob> job(
+    new WebAudioDecodeJob(contentType, this,
+                          promise, successCallback, failureCallback));
+  AsyncDecodeWebAudio(contentType.get(), data, length, *job);
+  // Transfer the ownership to mDecodeJobs
+  mDecodeJobs.AppendElement(job.forget());
+
+  return promise.forget();
+}
+
+void
+AudioContext::RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob)
+{
+  mDecodeJobs.RemoveElement(aDecodeJob);
+}
+
+void
+AudioContext::RegisterActiveNode(AudioNode* aNode)
+{
+  if (!mIsShutDown) {
+    mActiveNodes.PutEntry(aNode);
+  }
+}
+
+void
+AudioContext::UnregisterActiveNode(AudioNode* aNode)
+{
+  mActiveNodes.RemoveEntry(aNode);
+}
+
+void
+AudioContext::UnregisterAudioBufferSourceNode(AudioBufferSourceNode* aNode)
+{
+  UpdatePannerSource();
+}
+
+void
+AudioContext::UnregisterPannerNode(PannerNode* aNode)
+{
+  mPannerNodes.RemoveEntry(aNode);
+  if (mListener) {
+    mListener->UnregisterPannerNode(aNode);
+  }
+}
+
+void
+AudioContext::UpdatePannerSource()
+{
+  for (auto iter = mPannerNodes.Iter(); !iter.Done(); iter.Next()) {
+    iter.Get()->GetKey()->FindConnectedSources();
+  }
+}
+
+uint32_t
+AudioContext::MaxChannelCount() const
+{
+  return mIsOffline ? mNumberOfChannels : CubebUtils::MaxNumberOfChannels();
+}
+
+uint32_t
+AudioContext::ActiveNodeCount() const
+{
+  return mActiveNodes.Count();
+}
+
+MediaStreamGraph*
+AudioContext::Graph() const
+{
+  return Destination()->Stream()->Graph();
+}
+
+MediaStream*
+AudioContext::DestinationStream() const
+{
+  if (Destination()) {
+    return Destination()->Stream();
+  }
+  return nullptr;
+}
+
+double
+AudioContext::CurrentTime() const
+{
+  MediaStream* stream = Destination()->Stream();
+  return stream->StreamTimeToSeconds(stream->GetCurrentTime());
+}
+
+void
+AudioContext::Shutdown()
+{
+  mIsShutDown = true;
+
+  if (!mIsOffline) {
+    ErrorResult dummy;
+    RefPtr<Promise> ignored = Close(dummy);
+  }
+
+  for (auto p : mPromiseGripArray) {
+    p->MaybeReject(NS_ERROR_DOM_INVALID_STATE_ERR);
+  }
+
+  mPromiseGripArray.Clear();
+
+  // Release references to active nodes.
+  // Active AudioNodes don't unregister in destructors, at which point the
+  // Node is already unregistered.
+  mActiveNodes.Clear();
+
+  // For offline contexts, we can destroy the MediaStreamGraph at this point.
+  if (mIsOffline && mDestination) {
+    mDestination->OfflineShutdown();
+  }
+}
+
+StateChangeTask::StateChangeTask(AudioContext* aAudioContext,
+                                 void* aPromise,
+                                 AudioContextState aNewState)
+  : mAudioContext(aAudioContext)
+  , mPromise(aPromise)
+  , mAudioNodeStream(nullptr)
+  , mNewState(aNewState)
+{
+  MOZ_ASSERT(NS_IsMainThread(),
+             "This constructor should be used from the main thread.");
+}
+
+StateChangeTask::StateChangeTask(AudioNodeStream* aStream,
+                                 void* aPromise,
+                                 AudioContextState aNewState)
+  : mAudioContext(nullptr)
+  , mPromise(aPromise)
+  , mAudioNodeStream(aStream)
+  , mNewState(aNewState)
+{
+  MOZ_ASSERT(!NS_IsMainThread(),
+             "This constructor should be used from the graph thread.");
+}
+
+NS_IMETHODIMP
+StateChangeTask::Run()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mAudioContext && !mAudioNodeStream) {
+    return NS_OK;
+  }
+  if (mAudioNodeStream) {
+    AudioNode* node = mAudioNodeStream->Engine()->NodeMainThread();
+    if (!node) {
+      return NS_OK;
+    }
+    mAudioContext = node->Context();
+    if (!mAudioContext) {
+      return NS_OK;
+    }
+  }
+
+  mAudioContext->OnStateChanged(mPromise, mNewState);
+  // We have can't call Release() on the AudioContext on the MSG thread, so we
+  // unref it here, on the main thread.
+  mAudioContext = nullptr;
+
+  return NS_OK;
+}
+
+/* This runnable allows to fire the "statechange" event */
+class OnStateChangeTask final : public Runnable
+{
+public:
+  explicit OnStateChangeTask(AudioContext* aAudioContext)
+    : mAudioContext(aAudioContext)
+  {}
+
+  NS_IMETHODIMP
+  Run() override
+  {
+    nsPIDOMWindowInner* parent = mAudioContext->GetParentObject();
+    if (!parent) {
+      return NS_ERROR_FAILURE;
+    }
+
+    nsIDocument* doc = parent->GetExtantDoc();
+    if (!doc) {
+      return NS_ERROR_FAILURE;
+    }
+
+    return nsContentUtils::DispatchTrustedEvent(doc,
+                                static_cast<DOMEventTargetHelper*>(mAudioContext),
+                                NS_LITERAL_STRING("statechange"),
+                                false, false);
+  }
+
+private:
+  RefPtr<AudioContext> mAudioContext;
+};
+
+void
+AudioContext::OnStateChanged(void* aPromise, AudioContextState aNewState)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // This can happen if close() was called right after creating the
+  // AudioContext, before the context has switched to "running".
+  if (mAudioContextState == AudioContextState::Closed &&
+      aNewState == AudioContextState::Running &&
+      !aPromise) {
+    return;
+  }
+
+  // This can happen if this is called in reaction to a
+  // MediaStreamGraph shutdown, and a AudioContext was being
+  // suspended at the same time, for example if a page was being
+  // closed.
+  if (mAudioContextState == AudioContextState::Closed &&
+      aNewState == AudioContextState::Suspended) {
+    return;
+  }
+
+#ifndef WIN32 // Bug 1170547
+#ifndef XP_MACOSX
+#ifdef DEBUG
+
+  if (!((mAudioContextState == AudioContextState::Suspended &&
+       aNewState == AudioContextState::Running)   ||
+      (mAudioContextState == AudioContextState::Running   &&
+       aNewState == AudioContextState::Suspended) ||
+      (mAudioContextState == AudioContextState::Running   &&
+       aNewState == AudioContextState::Closed)    ||
+      (mAudioContextState == AudioContextState::Suspended &&
+       aNewState == AudioContextState::Closed)    ||
+      (mAudioContextState == aNewState))) {
+    fprintf(stderr,
+            "Invalid transition: mAudioContextState: %d -> aNewState %d\n",
+            static_cast<int>(mAudioContextState), static_cast<int>(aNewState));
+    MOZ_ASSERT(false);
+  }
+
+#endif // DEBUG
+#endif // XP_MACOSX
+#endif // WIN32
+
+  MOZ_ASSERT(
+    mIsOffline || aPromise || aNewState == AudioContextState::Running,
+    "We should have a promise here if this is a real-time AudioContext."
+    "Or this is the first time we switch to \"running\".");
+
+  if (aPromise) {
+    Promise* promise = reinterpret_cast<Promise*>(aPromise);
+    // It is possible for the promise to have been removed from
+    // mPromiseGripArray if the cycle collector has severed our connections. DO
+    // NOT dereference the promise pointer in that case since it may point to
+    // already freed memory.
+    if (mPromiseGripArray.Contains(promise)) {
+      promise->MaybeResolveWithUndefined();
+      DebugOnly<bool> rv = mPromiseGripArray.RemoveElement(promise);
+      MOZ_ASSERT(rv, "Promise wasn't in the grip array?");
+    }
+  }
+
+  if (mAudioContextState != aNewState) {
+    RefPtr<OnStateChangeTask> onStateChangeTask =
+      new OnStateChangeTask(this);
+    NS_DispatchToMainThread(onStateChangeTask);
+  }
+
+  mAudioContextState = aNewState;
+}
+
+nsTArray<MediaStream*>
+AudioContext::GetAllStreams() const
+{
+  nsTArray<MediaStream*> streams;
+  for (auto iter = mAllNodes.ConstIter(); !iter.Done(); iter.Next()) {
+    MediaStream* s = iter.Get()->GetKey()->GetStream();
+    if (s) {
+      streams.AppendElement(s);
+    }
+  }
+  return streams;
+}
+
+already_AddRefed<Promise>
+AudioContext::Suspend(ErrorResult& aRv)
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+  RefPtr<Promise> promise;
+  promise = Promise::Create(parentObject, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  if (mIsOffline) {
+    promise->MaybeReject(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return promise.forget();
+  }
+
+  if (mAudioContextState == AudioContextState::Closed ||
+      mCloseCalled) {
+    promise->MaybeReject(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return promise.forget();
+  }
+
+  Destination()->Suspend();
+
+  mPromiseGripArray.AppendElement(promise);
+
+  nsTArray<MediaStream*> streams;
+  // If mSuspendCalled is true then we already suspended all our streams,
+  // so don't suspend them again (since suspend(); suspend(); resume(); should
+  // cancel both suspends). But we still need to do ApplyAudioContextOperation
+  // to ensure our new promise is resolved.
+  if (!mSuspendCalled) {
+    streams = GetAllStreams();
+  }
+  Graph()->ApplyAudioContextOperation(DestinationStream()->AsAudioNodeStream(),
+                                      streams,
+                                      AudioContextOperation::Suspend, promise);
+
+  mSuspendCalled = true;
+
+  return promise.forget();
+}
+
+already_AddRefed<Promise>
+AudioContext::Resume(ErrorResult& aRv)
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+  RefPtr<Promise> promise;
+  promise = Promise::Create(parentObject, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  if (mIsOffline) {
+    promise->MaybeReject(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return promise.forget();
+  }
+
+  if (mAudioContextState == AudioContextState::Closed ||
+      mCloseCalled) {
+    promise->MaybeReject(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return promise.forget();
+  }
+
+  Destination()->Resume();
+
+  nsTArray<MediaStream*> streams;
+  // If mSuspendCalled is false then we already resumed all our streams,
+  // so don't resume them again (since suspend(); resume(); resume(); should
+  // be OK). But we still need to do ApplyAudioContextOperation
+  // to ensure our new promise is resolved.
+  if (mSuspendCalled) {
+    streams = GetAllStreams();
+  }
+  mPromiseGripArray.AppendElement(promise);
+  Graph()->ApplyAudioContextOperation(DestinationStream()->AsAudioNodeStream(),
+                                      streams,
+                                      AudioContextOperation::Resume, promise);
+
+  mSuspendCalled = false;
+
+  return promise.forget();
+}
+
+already_AddRefed<Promise>
+AudioContext::Close(ErrorResult& aRv)
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+  RefPtr<Promise> promise;
+  promise = Promise::Create(parentObject, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  if (mIsOffline) {
+    promise->MaybeReject(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return promise.forget();
+  }
+
+  if (mAudioContextState == AudioContextState::Closed) {
+    promise->MaybeResolve(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return promise.forget();
+  }
+
+  if (Destination()) {
+    Destination()->DestroyAudioChannelAgent();
+  }
+
+  mPromiseGripArray.AppendElement(promise);
+
+  // This can be called when freeing a document, and the streams are dead at
+  // this point, so we need extra null-checks.
+  MediaStream* ds = DestinationStream();
+  if (ds) {
+    nsTArray<MediaStream*> streams;
+    // If mSuspendCalled or mCloseCalled are true then we already suspended
+    // all our streams, so don't suspend them again. But we still need to do
+    // ApplyAudioContextOperation to ensure our new promise is resolved.
+    if (!mSuspendCalled && !mCloseCalled) {
+      streams = GetAllStreams();
+    }
+    Graph()->ApplyAudioContextOperation(ds->AsAudioNodeStream(), streams,
+                                        AudioContextOperation::Close, promise);
+  }
+  mCloseCalled = true;
+
+  return promise.forget();
+}
+
+void
+AudioContext::RegisterNode(AudioNode* aNode)
+{
+  MOZ_ASSERT(!mAllNodes.Contains(aNode));
+  mAllNodes.PutEntry(aNode);
+}
+
+void
+AudioContext::UnregisterNode(AudioNode* aNode)
+{
+  MOZ_ASSERT(mAllNodes.Contains(aNode));
+  mAllNodes.RemoveEntry(aNode);
+}
+
+JSObject*
+AudioContext::GetGlobalJSObject() const
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+  if (!parentObject) {
+    return nullptr;
+  }
+
+  // This can also return null.
+  return parentObject->GetGlobalJSObject();
+}
+
+already_AddRefed<Promise>
+AudioContext::StartRendering(ErrorResult& aRv)
+{
+  nsCOMPtr<nsIGlobalObject> parentObject = do_QueryInterface(GetParentObject());
+
+  MOZ_ASSERT(mIsOffline, "This should only be called on OfflineAudioContext");
+  if (mIsStarted) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return nullptr;
+  }
+
+  mIsStarted = true;
+  RefPtr<Promise> promise = Promise::Create(parentObject, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  mDestination->StartRendering(promise);
+
+  OnStateChanged(nullptr, AudioContextState::Running);
+
+  return promise.forget();
+}
+
+unsigned long
+AudioContext::Length()
+{
+  MOZ_ASSERT(mIsOffline);
+  return mDestination->Length();
+}
+
+void
+AudioContext::Mute() const
+{
+  MOZ_ASSERT(!mIsOffline);
+  if (mDestination) {
+    mDestination->Mute();
+  }
+}
+
+void
+AudioContext::Unmute() const
+{
+  MOZ_ASSERT(!mIsOffline);
+  if (mDestination) {
+    mDestination->Unmute();
+  }
+}
+
+AudioChannel
+AudioContext::MozAudioChannelType() const
+{
+  return mDestination->MozAudioChannelType();
+}
+
+AudioChannel
+AudioContext::TestAudioChannelInAudioNodeStream()
+{
+  MediaStream* stream = mDestination->Stream();
+  MOZ_ASSERT(stream);
+
+  return stream->AudioChannelType();
+}
+
+size_t
+AudioContext::SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+  // AudioNodes are tracked separately because we do not want the AudioContext
+  // to track all of the AudioNodes it creates, so we wouldn't be able to
+  // traverse them from here.
+
+  size_t amount = aMallocSizeOf(this);
+  if (mListener) {
+    amount += mListener->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  amount += mDecodeJobs.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (uint32_t i = 0; i < mDecodeJobs.Length(); ++i) {
+    amount += mDecodeJobs[i]->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  amount += mActiveNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += mPannerNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+NS_IMETHODIMP
+AudioContext::CollectReports(nsIHandleReportCallback* aHandleReport,
+                             nsISupports* aData, bool aAnonymize)
+{
+  const nsLiteralCString
+    nodeDescription("Memory used by AudioNode DOM objects (Web Audio).");
+  for (auto iter = mAllNodes.ConstIter(); !iter.Done(); iter.Next()) {
+    AudioNode* node = iter.Get()->GetKey();
+    int64_t amount = node->SizeOfIncludingThis(MallocSizeOf);
+    nsPrintfCString domNodePath("explicit/webaudio/audio-node/%s/dom-nodes",
+                                node->NodeType());
+    aHandleReport->Callback(EmptyCString(), domNodePath, KIND_HEAP, UNITS_BYTES,
+                            amount, nodeDescription, aData);
+  }
+
+  int64_t amount = SizeOfIncludingThis(MallocSizeOf);
+  MOZ_COLLECT_REPORT(
+    "explicit/webaudio/audiocontext", KIND_HEAP, UNITS_BYTES, amount,
+    "Memory used by AudioContext objects (Web Audio).");
+
+  return NS_OK;
+}
+
+BasicWaveFormCache*
+AudioContext::GetBasicWaveFormCache()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  if (!mBasicWaveFormCache) {
+    mBasicWaveFormCache = new BasicWaveFormCache(SampleRate());
+  }
+  return mBasicWaveFormCache;
+}
+
+BasicWaveFormCache::BasicWaveFormCache(uint32_t aSampleRate)
+  : mSampleRate(aSampleRate)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+}
+BasicWaveFormCache::~BasicWaveFormCache()
+{ }
+
+WebCore::PeriodicWave*
+BasicWaveFormCache::GetBasicWaveForm(OscillatorType aType)
+{
+  MOZ_ASSERT(!NS_IsMainThread());
+  if (aType == OscillatorType::Sawtooth) {
+    if (!mSawtooth) {
+      mSawtooth = WebCore::PeriodicWave::createSawtooth(mSampleRate);
+    }
+    return mSawtooth;
+  } else if (aType == OscillatorType::Square) {
+    if (!mSquare) {
+      mSquare = WebCore::PeriodicWave::createSquare(mSampleRate);
+    }
+    return mSquare;
+  } else if (aType == OscillatorType::Triangle) {
+    if (!mTriangle) {
+      mTriangle = WebCore::PeriodicWave::createTriangle(mSampleRate);
+    }
+    return mTriangle;
+  } else {
+    MOZ_ASSERT(false, "Not reached");
+    return nullptr;
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioContext.h b/dom/media/webaudio/AudioContext.h
new file mode 100644
index 0000000000..069efa9863
--- /dev/null
+++ b/dom/media/webaudio/AudioContext.h
@@ -0,0 +1,382 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioContext_h_
+#define AudioContext_h_
+
+#include "mozilla/dom/AudioChannelBinding.h"
+#include "MediaBufferDecoder.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/dom/TypedArray.h"
+#include "nsCOMPtr.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsHashKeys.h"
+#include "nsTHashtable.h"
+#include "js/TypeDecls.h"
+#include "nsIMemoryReporter.h"
+
+// X11 has a #define for CurrentTime. Unbelievable :-(.
+// See dom/media/DOMMediaStream.h for more fun!
+#ifdef CurrentTime
+#undef CurrentTime
+#endif
+
+namespace WebCore {
+  class PeriodicWave;
+} // namespace WebCore
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+
+class DOMMediaStream;
+class ErrorResult;
+class MediaStream;
+class MediaStreamGraph;
+class AudioNodeStream;
+
+namespace dom {
+
+enum class AudioContextState : uint32_t;
+class AnalyserNode;
+class AudioBuffer;
+class AudioBufferSourceNode;
+class AudioDestinationNode;
+class AudioListener;
+class AudioNode;
+class BiquadFilterNode;
+class ChannelMergerNode;
+class ChannelSplitterNode;
+class ConstantSourceNode;
+class ConvolverNode;
+class DelayNode;
+class DynamicsCompressorNode;
+class GainNode;
+class GlobalObject;
+class HTMLMediaElement;
+class IIRFilterNode;
+class MediaElementAudioSourceNode;
+class MediaStreamAudioDestinationNode;
+class MediaStreamAudioSourceNode;
+class OscillatorNode;
+class PannerNode;
+class ScriptProcessorNode;
+class StereoPannerNode;
+class WaveShaperNode;
+class PeriodicWave;
+struct PeriodicWaveConstraints;
+class Promise;
+enum class OscillatorType : uint32_t;
+
+// This is addrefed by the OscillatorNodeEngine on the main thread
+// and then used from the MSG thread.
+// It can be released either from the graph thread or the main thread.
+class BasicWaveFormCache
+{
+public:
+  explicit BasicWaveFormCache(uint32_t aSampleRate);
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(BasicWaveFormCache)
+  WebCore::PeriodicWave* GetBasicWaveForm(OscillatorType aType);
+private:
+  ~BasicWaveFormCache();
+  RefPtr<WebCore::PeriodicWave> mSawtooth;
+  RefPtr<WebCore::PeriodicWave> mSquare;
+  RefPtr<WebCore::PeriodicWave> mTriangle;
+  uint32_t mSampleRate;
+};
+
+
+/* This runnable allows the MSG to notify the main thread when audio is actually
+ * flowing */
+class StateChangeTask final : public Runnable
+{
+public:
+  /* This constructor should be used when this event is sent from the main
+   * thread. */
+  StateChangeTask(AudioContext* aAudioContext, void* aPromise, AudioContextState aNewState);
+
+  /* This constructor should be used when this event is sent from the audio
+   * thread. */
+  StateChangeTask(AudioNodeStream* aStream, void* aPromise, AudioContextState aNewState);
+
+  NS_IMETHOD Run() override;
+
+private:
+  RefPtr<AudioContext> mAudioContext;
+  void* mPromise;
+  RefPtr<AudioNodeStream> mAudioNodeStream;
+  AudioContextState mNewState;
+};
+
+enum class AudioContextOperation { Suspend, Resume, Close };
+
+class AudioContext final : public DOMEventTargetHelper,
+                           public nsIMemoryReporter
+{
+  AudioContext(nsPIDOMWindowInner* aParentWindow,
+               bool aIsOffline,
+               AudioChannel aChannel,
+               uint32_t aNumberOfChannels = 0,
+               uint32_t aLength = 0,
+               float aSampleRate = 0.0f);
+  ~AudioContext();
+
+  nsresult Init();
+
+public:
+  typedef uint64_t AudioContextId;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioContext,
+                                           DOMEventTargetHelper)
+  MOZ_DEFINE_MALLOC_SIZE_OF(MallocSizeOf)
+
+  nsPIDOMWindowInner* GetParentObject() const
+  {
+    return GetOwner();
+  }
+
+  void Shutdown(); // idempotent
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  using DOMEventTargetHelper::DispatchTrustedEvent;
+
+  // Constructor for regular AudioContext
+  static already_AddRefed<AudioContext>
+  Constructor(const GlobalObject& aGlobal, ErrorResult& aRv);
+
+  // Constructor for regular AudioContext. A default audio channel is needed.
+  static already_AddRefed<AudioContext>
+  Constructor(const GlobalObject& aGlobal,
+              AudioChannel aChannel,
+              ErrorResult& aRv);
+
+  // Constructor for offline AudioContext
+  static already_AddRefed<AudioContext>
+  Constructor(const GlobalObject& aGlobal,
+              uint32_t aNumberOfChannels,
+              uint32_t aLength,
+              float aSampleRate,
+              ErrorResult& aRv);
+
+  // AudioContext methods
+
+  AudioDestinationNode* Destination() const
+  {
+    return mDestination;
+  }
+
+  float SampleRate() const
+  {
+    return mSampleRate;
+  }
+
+  bool ShouldSuspendNewStream() const { return mSuspendCalled; }
+
+  double CurrentTime() const;
+
+  AudioListener* Listener();
+
+  AudioContextState State() const { return mAudioContextState; }
+
+  // Those three methods return a promise to content, that is resolved when an
+  // (possibly long) operation is completed on the MSG (and possibly other)
+  // thread(s). To avoid having to match the calls and asychronous result when
+  // the operation is completed, we keep a reference to the promises on the main
+  // thread, and then send the promises pointers down the MSG thread, as a void*
+  // (to make it very clear that the pointer is to merely be treated as an ID).
+  // When back on the main thread, we can resolve or reject the promise, by
+  // casting it back to a `Promise*` while asserting we're back on the main
+  // thread and removing the reference we added.
+  already_AddRefed<Promise> Suspend(ErrorResult& aRv);
+  already_AddRefed<Promise> Resume(ErrorResult& aRv);
+  already_AddRefed<Promise> Close(ErrorResult& aRv);
+  IMPL_EVENT_HANDLER(statechange)
+
+  already_AddRefed<AudioBufferSourceNode> CreateBufferSource(ErrorResult& aRv);
+
+  already_AddRefed<ConstantSourceNode> CreateConstantSource(ErrorResult& aRv);
+
+  already_AddRefed<AudioBuffer>
+  CreateBuffer(uint32_t aNumberOfChannels, uint32_t aLength, float aSampleRate,
+               ErrorResult& aRv);
+
+  already_AddRefed<MediaStreamAudioDestinationNode>
+  CreateMediaStreamDestination(ErrorResult& aRv);
+
+  already_AddRefed<ScriptProcessorNode>
+  CreateScriptProcessor(uint32_t aBufferSize,
+                        uint32_t aNumberOfInputChannels,
+                        uint32_t aNumberOfOutputChannels,
+                        ErrorResult& aRv);
+
+  already_AddRefed<StereoPannerNode>
+  CreateStereoPanner(ErrorResult& aRv);
+
+  already_AddRefed<AnalyserNode>
+  CreateAnalyser(ErrorResult& aRv);
+
+  already_AddRefed<GainNode>
+  CreateGain(ErrorResult& aRv);
+
+  already_AddRefed<WaveShaperNode>
+  CreateWaveShaper(ErrorResult& aRv);
+
+  already_AddRefed<MediaElementAudioSourceNode>
+  CreateMediaElementSource(HTMLMediaElement& aMediaElement, ErrorResult& aRv);
+  already_AddRefed<MediaStreamAudioSourceNode>
+  CreateMediaStreamSource(DOMMediaStream& aMediaStream, ErrorResult& aRv);
+
+  already_AddRefed<DelayNode>
+  CreateDelay(double aMaxDelayTime, ErrorResult& aRv);
+
+  already_AddRefed<PannerNode>
+  CreatePanner(ErrorResult& aRv);
+
+  already_AddRefed<ConvolverNode>
+  CreateConvolver(ErrorResult& aRv);
+
+  already_AddRefed<ChannelSplitterNode>
+  CreateChannelSplitter(uint32_t aNumberOfOutputs, ErrorResult& aRv);
+
+  already_AddRefed<ChannelMergerNode>
+  CreateChannelMerger(uint32_t aNumberOfInputs, ErrorResult& aRv);
+
+  already_AddRefed<DynamicsCompressorNode>
+  CreateDynamicsCompressor(ErrorResult& aRv);
+
+  already_AddRefed<BiquadFilterNode>
+  CreateBiquadFilter(ErrorResult& aRv);
+
+  already_AddRefed<IIRFilterNode>
+  CreateIIRFilter(const mozilla::dom::binding_detail::AutoSequence<double>& aFeedforward,
+                  const mozilla::dom::binding_detail::AutoSequence<double>& aFeedback,
+                  mozilla::ErrorResult& aRv);
+
+  already_AddRefed<OscillatorNode>
+  CreateOscillator(ErrorResult& aRv);
+
+  already_AddRefed<PeriodicWave>
+  CreatePeriodicWave(const Float32Array& aRealData, const Float32Array& aImagData,
+                     const PeriodicWaveConstraints& aConstraints,
+                     ErrorResult& aRv);
+
+  already_AddRefed<Promise>
+  DecodeAudioData(const ArrayBuffer& aBuffer,
+                  const Optional<OwningNonNull<DecodeSuccessCallback> >& aSuccessCallback,
+                  const Optional<OwningNonNull<DecodeErrorCallback> >& aFailureCallback,
+                  ErrorResult& aRv);
+
+  // OfflineAudioContext methods
+  already_AddRefed<Promise> StartRendering(ErrorResult& aRv);
+  IMPL_EVENT_HANDLER(complete)
+  unsigned long Length();
+
+  bool IsOffline() const { return mIsOffline; }
+
+  MediaStreamGraph* Graph() const;
+  MediaStream* DestinationStream() const;
+
+  // Nodes register here if they will produce sound even if they have silent
+  // or no input connections.  The AudioContext will keep registered nodes
+  // alive until the context is collected.  This takes care of "playing"
+  // references and "tail-time" references.
+  void RegisterActiveNode(AudioNode* aNode);
+  // Nodes unregister when they have finished producing sound for the
+  // foreseeable future.
+  // Do NOT call UnregisterActiveNode from an AudioNode destructor.
+  // If the destructor is called, then the Node has already been unregistered.
+  // The destructor may be called during hashtable enumeration, during which
+  // unregistering would not be safe.
+  void UnregisterActiveNode(AudioNode* aNode);
+
+  void UnregisterAudioBufferSourceNode(AudioBufferSourceNode* aNode);
+  void UnregisterPannerNode(PannerNode* aNode);
+  void UpdatePannerSource();
+
+  uint32_t MaxChannelCount() const;
+
+  uint32_t ActiveNodeCount() const;
+
+  void Mute() const;
+  void Unmute() const;
+
+  JSObject* GetGlobalJSObject() const;
+
+  AudioChannel MozAudioChannelType() const;
+
+  AudioChannel TestAudioChannelInAudioNodeStream();
+
+  void RegisterNode(AudioNode* aNode);
+  void UnregisterNode(AudioNode* aNode);
+
+  void OnStateChanged(void* aPromise, AudioContextState aNewState);
+
+  BasicWaveFormCache* GetBasicWaveFormCache();
+
+  IMPL_EVENT_HANDLER(mozinterruptbegin)
+  IMPL_EVENT_HANDLER(mozinterruptend)
+
+private:
+  void DisconnectFromWindow();
+  void RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob);
+  void ShutdownDecoder();
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+  NS_DECL_NSIMEMORYREPORTER
+
+  friend struct ::mozilla::WebAudioDecodeJob;
+
+  bool CheckClosed(ErrorResult& aRv);
+
+  nsTArray<MediaStream*> GetAllStreams() const;
+
+private:
+  // Each AudioContext has an id, that is passed down the MediaStreams that
+  // back the AudioNodes, so we can easily compute the set of all the
+  // MediaStreams for a given context, on the MediasStreamGraph side.
+  const AudioContextId mId;
+  // Note that it's important for mSampleRate to be initialized before
+  // mDestination, as mDestination's constructor needs to access it!
+  const float mSampleRate;
+  AudioContextState mAudioContextState;
+  RefPtr<AudioDestinationNode> mDestination;
+  RefPtr<AudioListener> mListener;
+  nsTArray<RefPtr<WebAudioDecodeJob> > mDecodeJobs;
+  // This array is used to keep the suspend/resume/close promises alive until
+  // they are resolved, so we can safely pass them accross threads.
+  nsTArray<RefPtr<Promise>> mPromiseGripArray;
+  // See RegisterActiveNode.  These will keep the AudioContext alive while it
+  // is rendering and the window remains alive.
+  nsTHashtable<nsRefPtrHashKey<AudioNode> > mActiveNodes;
+  // Raw (non-owning) references to all AudioNodes for this AudioContext.
+  nsTHashtable<nsPtrHashKey<AudioNode> > mAllNodes;
+  // Hashsets containing all the PannerNodes, to compute the doppler shift.
+  // These are weak pointers.
+  nsTHashtable<nsPtrHashKey<PannerNode> > mPannerNodes;
+  // Cache to avoid recomputing basic waveforms all the time.
+  RefPtr<BasicWaveFormCache> mBasicWaveFormCache;
+  // Number of channels passed in the OfflineAudioContext ctor.
+  uint32_t mNumberOfChannels;
+  bool mIsOffline;
+  bool mIsStarted;
+  bool mIsShutDown;
+  // Close has been called, reject suspend and resume call.
+  bool mCloseCalled;
+  // Suspend has been called with no following resume.
+  bool mSuspendCalled;
+};
+
+static const dom::AudioContext::AudioContextId NO_AUDIO_CONTEXT = 0;
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioDestinationNode.cpp b/dom/media/webaudio/AudioDestinationNode.cpp
new file mode 100644
index 0000000000..29a9de736c
--- /dev/null
+++ b/dom/media/webaudio/AudioDestinationNode.cpp
@@ -0,0 +1,680 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioDestinationNode.h"
+#include "AlignmentUtils.h"
+#include "AudioContext.h"
+#include "mozilla/dom/AudioDestinationNodeBinding.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "mozilla/Services.h"
+#include "AudioChannelAgent.h"
+#include "AudioChannelService.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "MediaStreamGraph.h"
+#include "OfflineAudioCompletionEvent.h"
+#include "nsContentUtils.h"
+#include "nsIInterfaceRequestorUtils.h"
+#include "nsIDocShell.h"
+#include "nsIPermissionManager.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsServiceManagerUtils.h"
+#include "mozilla/dom/Promise.h"
+
+namespace mozilla {
+namespace dom {
+
+static uint8_t gWebAudioOutputKey;
+
+class OfflineDestinationNodeEngine final : public AudioNodeEngine
+{
+public:
+  OfflineDestinationNodeEngine(AudioDestinationNode* aNode,
+                               uint32_t aNumberOfChannels,
+                               uint32_t aLength,
+                               float aSampleRate)
+    : AudioNodeEngine(aNode)
+    , mWriteIndex(0)
+    , mNumberOfChannels(aNumberOfChannels)
+    , mLength(aLength)
+    , mSampleRate(aSampleRate)
+    , mBufferAllocated(false)
+  {
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    // Do this just for the sake of political correctness; this output
+    // will not go anywhere.
+    *aOutput = aInput;
+
+    // The output buffer is allocated lazily, on the rendering thread, when
+    // non-null input is received.
+    if (!mBufferAllocated && !aInput.IsNull()) {
+      // These allocations might fail if content provides a huge number of
+      // channels or size, but it's OK since we'll deal with the failure
+      // gracefully.
+      mBuffer = ThreadSharedFloatArrayBufferList::
+        Create(mNumberOfChannels, mLength, fallible);
+      if (mBuffer && mWriteIndex) {
+        // Zero leading for any null chunks that were skipped.
+        for (uint32_t i = 0; i < mNumberOfChannels; ++i) {
+          float* channelData = mBuffer->GetDataForWrite(i);
+          PodZero(channelData, mWriteIndex);
+        }
+      }
+
+      mBufferAllocated = true;
+    }
+
+    // Skip copying if there is no buffer.
+    uint32_t outputChannelCount = mBuffer ? mNumberOfChannels : 0;
+
+    // Record our input buffer
+    MOZ_ASSERT(mWriteIndex < mLength, "How did this happen?");
+    const uint32_t duration = std::min(WEBAUDIO_BLOCK_SIZE, mLength - mWriteIndex);
+    const uint32_t inputChannelCount = aInput.ChannelCount();
+    for (uint32_t i = 0; i < outputChannelCount; ++i) {
+      float* outputData = mBuffer->GetDataForWrite(i) + mWriteIndex;
+      if (aInput.IsNull() || i >= inputChannelCount) {
+        PodZero(outputData, duration);
+      } else {
+        const float* inputBuffer = static_cast<const float*>(aInput.mChannelData[i]);
+        if (duration == WEBAUDIO_BLOCK_SIZE && IS_ALIGNED16(inputBuffer)) {
+          // Use the optimized version of the copy with scale operation
+          AudioBlockCopyChannelWithScale(inputBuffer, aInput.mVolume,
+                                         outputData);
+        } else {
+          if (aInput.mVolume == 1.0f) {
+            PodCopy(outputData, inputBuffer, duration);
+          } else {
+            for (uint32_t j = 0; j < duration; ++j) {
+              outputData[j] = aInput.mVolume * inputBuffer[j];
+            }
+          }
+        }
+      }
+    }
+    mWriteIndex += duration;
+
+    if (mWriteIndex >= mLength) {
+      NS_ASSERTION(mWriteIndex == mLength, "Overshot length");
+      // Go to finished state. When the graph's current time eventually reaches
+      // the end of the stream, then the main thread will be notified and we'll
+      // shut down the AudioContext.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // Keep processing to track stream time, which is used for all timelines
+    // associated with the same AudioContext.
+    return true;
+  }
+
+
+  class OnCompleteTask final : public Runnable
+  {
+  public:
+    OnCompleteTask(AudioContext* aAudioContext, AudioBuffer* aRenderedBuffer)
+      : mAudioContext(aAudioContext)
+      , mRenderedBuffer(aRenderedBuffer)
+    {}
+
+    NS_IMETHOD Run() override
+    {
+      RefPtr<OfflineAudioCompletionEvent> event =
+          new OfflineAudioCompletionEvent(mAudioContext, nullptr, nullptr);
+      event->InitEvent(mRenderedBuffer);
+      mAudioContext->DispatchTrustedEvent(event);
+
+      return NS_OK;
+    }
+  private:
+    RefPtr<AudioContext> mAudioContext;
+    RefPtr<AudioBuffer> mRenderedBuffer;
+  };
+
+  void FireOfflineCompletionEvent(AudioDestinationNode* aNode)
+  {
+    AudioContext* context = aNode->Context();
+    context->Shutdown();
+    // Shutdown drops self reference, but the context is still referenced by aNode,
+    // which is strongly referenced by the runnable that called
+    // AudioDestinationNode::FireOfflineCompletionEvent.
+
+    // Create the input buffer
+    ErrorResult rv;
+    RefPtr<AudioBuffer> renderedBuffer =
+      AudioBuffer::Create(context, mNumberOfChannels, mLength, mSampleRate,
+                          mBuffer.forget(), rv);
+    if (rv.Failed()) {
+      rv.SuppressException();
+      return;
+    }
+
+    aNode->ResolvePromise(renderedBuffer);
+
+    RefPtr<OnCompleteTask> onCompleteTask =
+      new OnCompleteTask(context, renderedBuffer);
+    NS_DispatchToMainThread(onCompleteTask);
+
+    context->OnStateChanged(nullptr, AudioContextState::Closed);
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    if (mBuffer) {
+      amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  // The input to the destination node is recorded in mBuffer.
+  // When this buffer fills up with mLength frames, the buffered input is sent
+  // to the main thread in order to dispatch OfflineAudioCompletionEvent.
+  RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  // An index representing the next offset in mBuffer to be written to.
+  uint32_t mWriteIndex;
+  uint32_t mNumberOfChannels;
+  // How many frames the OfflineAudioContext intends to produce.
+  uint32_t mLength;
+  float mSampleRate;
+  bool mBufferAllocated;
+};
+
+class InputMutedRunnable final : public Runnable
+{
+public:
+  InputMutedRunnable(AudioNodeStream* aStream,
+                     bool aInputMuted)
+    : mStream(aStream)
+    , mInputMuted(aInputMuted)
+  {
+  }
+
+  NS_IMETHOD Run() override
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    RefPtr<AudioNode> node = mStream->Engine()->NodeMainThread();
+
+    if (node) {
+      RefPtr<AudioDestinationNode> destinationNode =
+        static_cast<AudioDestinationNode*>(node.get());
+      destinationNode->InputMuted(mInputMuted);
+    }
+    return NS_OK;
+  }
+
+private:
+  RefPtr<AudioNodeStream> mStream;
+  bool mInputMuted;
+};
+
+class DestinationNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit DestinationNodeEngine(AudioDestinationNode* aNode)
+    : AudioNodeEngine(aNode)
+    , mVolume(1.0f)
+    , mLastInputMuted(true)
+    , mSuspended(false)
+  {
+    MOZ_ASSERT(aNode);
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    *aOutput = aInput;
+    aOutput->mVolume *= mVolume;
+
+    if (mSuspended) {
+      return;
+    }
+
+    bool newInputMuted = aInput.IsNull() || aInput.IsMuted();
+    if (newInputMuted != mLastInputMuted) {
+      mLastInputMuted = newInputMuted;
+
+      RefPtr<InputMutedRunnable> runnable =
+        new InputMutedRunnable(aStream, newInputMuted);
+      aStream->Graph()->
+        DispatchToMainThreadAfterStreamStateUpdate(runnable.forget());
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // Keep processing to track stream time, which is used for all timelines
+    // associated with the same AudioContext.  If there are no other engines
+    // for the AudioContext, then this could return false to suspend the
+    // stream, but the stream is blocked anyway through
+    // AudioDestinationNode::SetIsOnlyNodeForContext().
+    return true;
+  }
+
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override
+  {
+    if (aIndex == VOLUME) {
+      mVolume = aParam;
+    }
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    if (aIndex == SUSPENDED) {
+      mSuspended = !!aParam;
+      if (mSuspended) {
+        mLastInputMuted = true;
+      }
+    }
+  }
+
+  enum Parameters {
+    VOLUME,
+    SUSPENDED,
+  };
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  float mVolume;
+  bool mLastInputMuted;
+  bool mSuspended;
+};
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioDestinationNode, AudioNode,
+                                   mAudioChannelAgent,
+                                   mOfflineRenderingPromise)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioDestinationNode)
+  NS_INTERFACE_MAP_ENTRY(nsIAudioChannelAgentCallback)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(AudioDestinationNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(AudioDestinationNode, AudioNode)
+
+AudioDestinationNode::AudioDestinationNode(AudioContext* aContext,
+                                           bool aIsOffline,
+                                           AudioChannel aChannel,
+                                           uint32_t aNumberOfChannels,
+                                           uint32_t aLength, float aSampleRate)
+  : AudioNode(aContext, aIsOffline ? aNumberOfChannels : 2,
+              ChannelCountMode::Explicit, ChannelInterpretation::Speakers)
+  , mFramesToProduce(aLength)
+  , mAudioChannel(AudioChannel::Normal)
+  , mIsOffline(aIsOffline)
+  , mAudioChannelSuspended(false)
+  , mCaptured(false)
+{
+  MediaStreamGraph* graph = aIsOffline ?
+                            MediaStreamGraph::CreateNonRealtimeInstance(aSampleRate) :
+                            MediaStreamGraph::GetInstance(MediaStreamGraph::AUDIO_THREAD_DRIVER, aChannel);
+  AudioNodeEngine* engine = aIsOffline ?
+                            new OfflineDestinationNodeEngine(this, aNumberOfChannels,
+                                                             aLength, aSampleRate) :
+                            static_cast<AudioNodeEngine*>(new DestinationNodeEngine(this));
+
+  AudioNodeStream::Flags flags =
+    AudioNodeStream::NEED_MAIN_THREAD_CURRENT_TIME |
+    AudioNodeStream::NEED_MAIN_THREAD_FINISHED |
+    AudioNodeStream::EXTERNAL_OUTPUT;
+  mStream = AudioNodeStream::Create(aContext, engine, flags, graph);
+  mStream->AddMainThreadListener(this);
+  mStream->AddAudioOutput(&gWebAudioOutputKey);
+
+  if (!aIsOffline) {
+    graph->NotifyWhenGraphStarted(mStream);
+  }
+
+  if (aChannel != AudioChannel::Normal) {
+    ErrorResult rv;
+    SetMozAudioChannelType(aChannel, rv);
+  }
+}
+
+AudioDestinationNode::~AudioDestinationNode()
+{
+}
+
+size_t
+AudioDestinationNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  // Might be useful in the future:
+  // - mAudioChannelAgent
+  return amount;
+}
+
+size_t
+AudioDestinationNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void
+AudioDestinationNode::DestroyAudioChannelAgent()
+{
+  if (mAudioChannelAgent && !Context()->IsOffline()) {
+    mAudioChannelAgent->NotifyStoppedPlaying();
+    mAudioChannelAgent = nullptr;
+  }
+}
+
+void
+AudioDestinationNode::DestroyMediaStream()
+{
+  DestroyAudioChannelAgent();
+
+  if (!mStream)
+    return;
+
+  mStream->RemoveMainThreadListener(this);
+  MediaStreamGraph* graph = mStream->Graph();
+  if (graph->IsNonRealtime()) {
+    MediaStreamGraph::DestroyNonRealtimeInstance(graph);
+  }
+  AudioNode::DestroyMediaStream();
+}
+
+void
+AudioDestinationNode::NotifyMainThreadStreamFinished()
+{
+  MOZ_ASSERT(mStream->IsFinished());
+
+  if (mIsOffline) {
+    NS_DispatchToCurrentThread(NewRunnableMethod(this,
+                                                 &AudioDestinationNode::FireOfflineCompletionEvent));
+  }
+}
+
+void
+AudioDestinationNode::FireOfflineCompletionEvent()
+{
+  OfflineDestinationNodeEngine* engine =
+    static_cast<OfflineDestinationNodeEngine*>(Stream()->Engine());
+  engine->FireOfflineCompletionEvent(this);
+}
+
+void
+AudioDestinationNode::ResolvePromise(AudioBuffer* aRenderedBuffer)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mIsOffline);
+  mOfflineRenderingPromise->MaybeResolve(aRenderedBuffer);
+}
+
+uint32_t
+AudioDestinationNode::MaxChannelCount() const
+{
+  return Context()->MaxChannelCount();
+}
+
+void
+AudioDestinationNode::SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv)
+{
+  if (aChannelCount > MaxChannelCount()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  AudioNode::SetChannelCount(aChannelCount, aRv);
+}
+
+void
+AudioDestinationNode::Mute()
+{
+  MOZ_ASSERT(Context() && !Context()->IsOffline());
+  SendDoubleParameterToStream(DestinationNodeEngine::VOLUME, 0.0f);
+}
+
+void
+AudioDestinationNode::Unmute()
+{
+  MOZ_ASSERT(Context() && !Context()->IsOffline());
+  SendDoubleParameterToStream(DestinationNodeEngine::VOLUME, 1.0f);
+}
+
+void
+AudioDestinationNode::Suspend()
+{
+  DestroyAudioChannelAgent();
+  SendInt32ParameterToStream(DestinationNodeEngine::SUSPENDED, 1);
+}
+
+void
+AudioDestinationNode::Resume()
+{
+  CreateAudioChannelAgent();
+  SendInt32ParameterToStream(DestinationNodeEngine::SUSPENDED, 0);
+}
+
+void
+AudioDestinationNode::OfflineShutdown()
+{
+  MOZ_ASSERT(Context() && Context()->IsOffline(),
+             "Should only be called on a valid OfflineAudioContext");
+
+  MediaStreamGraph::DestroyNonRealtimeInstance(mStream->Graph());
+  mOfflineRenderingRef.Drop(this);
+}
+
+JSObject*
+AudioDestinationNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioDestinationNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+AudioDestinationNode::StartRendering(Promise* aPromise)
+{
+  mOfflineRenderingPromise = aPromise;
+  mOfflineRenderingRef.Take(this);
+  mStream->Graph()->StartNonRealtimeProcessing(mFramesToProduce);
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowVolumeChanged(float aVolume, bool aMuted)
+{
+  if (!mStream) {
+    return NS_OK;
+  }
+
+  float volume = aMuted ? 0.0 : aVolume;
+  mStream->SetAudioOutputVolume(&gWebAudioOutputKey, volume);
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowSuspendChanged(nsSuspendedTypes aSuspend)
+{
+  if (!mStream) {
+    return NS_OK;
+  }
+
+  bool suspended = (aSuspend != nsISuspendedTypes::NONE_SUSPENDED);
+  if (mAudioChannelSuspended == suspended) {
+    return NS_OK;
+  }
+
+  mAudioChannelSuspended = suspended;
+  Context()->DispatchTrustedEvent(!suspended ?
+    NS_LITERAL_STRING("mozinterruptend") :
+    NS_LITERAL_STRING("mozinterruptbegin"));
+
+  DisabledTrackMode disabledMode = suspended ? DisabledTrackMode::SILENCE_BLACK
+                                             : DisabledTrackMode::ENABLED;
+  mStream->SetTrackEnabled(AudioNodeStream::AUDIO_TRACK, disabledMode);
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowAudioCaptureChanged(bool aCapture)
+{
+  MOZ_ASSERT(mAudioChannelAgent);
+
+  if (!mStream || Context()->IsOffline()) {
+    return NS_OK;
+  }
+
+  nsCOMPtr<nsPIDOMWindowInner> ownerWindow = GetOwner();
+  if (!ownerWindow) {
+    return NS_OK;
+  }
+
+  if (aCapture != mCaptured) {
+    if (aCapture) {
+      nsCOMPtr<nsPIDOMWindowInner> window = Context()->GetParentObject();
+      uint64_t id = window->WindowID();
+      mCaptureStreamPort =
+        mStream->Graph()->ConnectToCaptureStream(id, mStream);
+    } else {
+      mCaptureStreamPort->Destroy();
+    }
+    mCaptured = aCapture;
+  }
+
+  return NS_OK;
+}
+
+AudioChannel
+AudioDestinationNode::MozAudioChannelType() const
+{
+  return mAudioChannel;
+}
+
+void
+AudioDestinationNode::SetMozAudioChannelType(AudioChannel aValue, ErrorResult& aRv)
+{
+  if (Context()->IsOffline()) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+
+  if (aValue != mAudioChannel &&
+      CheckAudioChannelPermissions(aValue)) {
+    mAudioChannel = aValue;
+
+    if (mStream) {
+      mStream->SetAudioChannelType(mAudioChannel);
+    }
+
+    if (mAudioChannelAgent) {
+      CreateAudioChannelAgent();
+    }
+  }
+}
+
+bool
+AudioDestinationNode::CheckAudioChannelPermissions(AudioChannel aValue)
+{
+  // Only normal channel doesn't need permission.
+  if (aValue == AudioChannel::Normal) {
+    return true;
+  }
+
+  // Maybe this audio channel is equal to the default one.
+  if (aValue == AudioChannelService::GetDefaultAudioChannel()) {
+    return true;
+  }
+
+  nsCOMPtr<nsIPermissionManager> permissionManager =
+    services::GetPermissionManager();
+  if (!permissionManager) {
+    return false;
+  }
+
+  nsCOMPtr<nsIScriptObjectPrincipal> sop = do_QueryInterface(GetOwner());
+  NS_ASSERTION(sop, "Window didn't QI to nsIScriptObjectPrincipal!");
+  nsCOMPtr<nsIPrincipal> principal = sop->GetPrincipal();
+
+  uint32_t perm = nsIPermissionManager::UNKNOWN_ACTION;
+
+  nsCString channel;
+  channel.AssignASCII(AudioChannelValues::strings[uint32_t(aValue)].value,
+                      AudioChannelValues::strings[uint32_t(aValue)].length);
+  permissionManager->TestExactPermissionFromPrincipal(principal,
+    nsCString(NS_LITERAL_CSTRING("audio-channel-") + channel).get(),
+    &perm);
+
+  return perm == nsIPermissionManager::ALLOW_ACTION;
+}
+
+nsresult
+AudioDestinationNode::CreateAudioChannelAgent()
+{
+  if (mIsOffline) {
+    return NS_OK;
+  }
+
+  nsresult rv = NS_OK;
+  if (mAudioChannelAgent) {
+    rv = mAudioChannelAgent->NotifyStoppedPlaying();
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      return rv;
+    }
+  }
+
+  mAudioChannelAgent = new AudioChannelAgent();
+  rv = mAudioChannelAgent->InitWithWeakCallback(GetOwner(),
+                                           static_cast<int32_t>(mAudioChannel),
+                                           this);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  return NS_OK;
+}
+
+void
+AudioDestinationNode::InputMuted(bool aMuted)
+{
+  MOZ_ASSERT(Context() && !Context()->IsOffline());
+
+  if (!mAudioChannelAgent) {
+    if (aMuted) {
+      return;
+    }
+    CreateAudioChannelAgent();
+  }
+
+  if (aMuted) {
+    mAudioChannelAgent->NotifyStoppedPlaying();
+    return;
+  }
+
+  AudioPlaybackConfig config;
+  nsresult rv = mAudioChannelAgent->NotifyStartedPlaying(&config,
+                                                         AudioChannelService::AudibleState::eAudible);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return;
+  }
+
+  WindowVolumeChanged(config.mVolume, config.mMuted);
+  WindowSuspendChanged(config.mSuspend);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioDestinationNode.h b/dom/media/webaudio/AudioDestinationNode.h
new file mode 100644
index 0000000000..cf0db78627
--- /dev/null
+++ b/dom/media/webaudio/AudioDestinationNode.h
@@ -0,0 +1,115 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioDestinationNode_h_
+#define AudioDestinationNode_h_
+
+#include "mozilla/dom/AudioChannelBinding.h"
+#include "AudioNode.h"
+#include "nsIAudioChannelAgent.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class AudioDestinationNode final : public AudioNode
+                                 , public nsIAudioChannelAgentCallback
+                                 , public MainThreadMediaStreamListener
+{
+public:
+  // This node type knows what MediaStreamGraph to use based on
+  // whether it's in offline mode.
+  AudioDestinationNode(AudioContext* aContext,
+                       bool aIsOffline,
+                       AudioChannel aChannel = AudioChannel::Normal,
+                       uint32_t aNumberOfChannels = 0,
+                       uint32_t aLength = 0,
+                       float aSampleRate = 0.0f);
+
+  void DestroyMediaStream() override;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioDestinationNode, AudioNode)
+  NS_DECL_NSIAUDIOCHANNELAGENTCALLBACK
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const final override
+  {
+    return 0;
+  }
+
+  uint32_t MaxChannelCount() const;
+  void SetChannelCount(uint32_t aChannelCount,
+                       ErrorResult& aRv) override;
+
+  // Returns the stream or null after unlink.
+  AudioNodeStream* Stream() { return mStream; }
+
+  void Mute();
+  void Unmute();
+
+  void Suspend();
+  void Resume();
+
+  void StartRendering(Promise* aPromise);
+
+  void OfflineShutdown();
+
+  AudioChannel MozAudioChannelType() const;
+
+  void NotifyMainThreadStreamFinished() override;
+  void FireOfflineCompletionEvent();
+
+  nsresult CreateAudioChannelAgent();
+  void DestroyAudioChannelAgent();
+
+  const char* NodeType() const override
+  {
+    return "AudioDestinationNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void InputMuted(bool aInputMuted);
+  void ResolvePromise(AudioBuffer* aRenderedBuffer);
+
+  unsigned long Length()
+  {
+    MOZ_ASSERT(mIsOffline);
+    return mFramesToProduce;
+  }
+
+protected:
+  virtual ~AudioDestinationNode();
+
+private:
+  void SetMozAudioChannelType(AudioChannel aValue, ErrorResult& aRv);
+  bool CheckAudioChannelPermissions(AudioChannel aValue);
+
+  SelfReference<AudioDestinationNode> mOfflineRenderingRef;
+  uint32_t mFramesToProduce;
+
+  nsCOMPtr<nsIAudioChannelAgent> mAudioChannelAgent;
+  RefPtr<MediaInputPort> mCaptureStreamPort;
+
+  RefPtr<Promise> mOfflineRenderingPromise;
+
+  // Audio Channel Type.
+  AudioChannel mAudioChannel;
+  bool mIsOffline;
+  bool mAudioChannelSuspended;
+
+  bool mCaptured;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioEventTimeline.cpp b/dom/media/webaudio/AudioEventTimeline.cpp
new file mode 100644
index 0000000000..a6a7bbf660
--- /dev/null
+++ b/dom/media/webaudio/AudioEventTimeline.cpp
@@ -0,0 +1,315 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioEventTimeline.h"
+
+#include "mozilla/ErrorResult.h"
+
+static float LinearInterpolate(double t0, float v0, double t1, float v1, double t)
+{
+  return v0 + (v1 - v0) * ((t - t0) / (t1 - t0));
+}
+
+static float ExponentialInterpolate(double t0, float v0, double t1, float v1, double t)
+{
+  return v0 * powf(v1 / v0, (t - t0) / (t1 - t0));
+}
+
+static float ExponentialApproach(double t0, double v0, float v1, double timeConstant, double t)
+{
+  if (!mozilla::dom::WebAudioUtils::FuzzyEqual(timeConstant, 0.0)) {
+    return v1 + (v0 - v1) * expf(-(t - t0) / timeConstant);
+  } else {
+    return v1;
+  }
+}
+
+static float ExtractValueFromCurve(double startTime, float* aCurve, uint32_t aCurveLength, double duration, double t)
+{
+  if (t >= startTime + duration) {
+    // After the duration, return the last curve value
+    return aCurve[aCurveLength - 1];
+  }
+  double ratio = std::max((t - startTime) / duration, 0.0);
+  if (ratio >= 1.0) {
+    return aCurve[aCurveLength - 1];
+  }
+  uint32_t current = uint32_t(floor((aCurveLength - 1) * ratio));
+  uint32_t next = current + 1;
+  double step = duration / double(aCurveLength - 1);
+  if (next < aCurveLength) {
+    double t0 = current * step;
+    double t1 = next * step;
+    return LinearInterpolate(t0, aCurve[current], t1, aCurve[next], t - startTime);
+  } else {
+    return aCurve[current];
+  }
+}
+
+namespace mozilla {
+namespace dom {
+
+// This method computes the AudioParam value at a given time based on the event timeline
+template<class TimeType> void
+AudioEventTimeline::GetValuesAtTimeHelper(TimeType aTime, float* aBuffer,
+                                          const size_t aSize)
+{
+  MOZ_ASSERT(aBuffer);
+  MOZ_ASSERT(aSize);
+
+  auto TimeOf = [](const AudioTimelineEvent& aEvent) -> TimeType {
+    return aEvent.template Time<TimeType>();
+  };
+
+  size_t eventIndex = 0;
+  const AudioTimelineEvent* previous = nullptr;
+
+  // Let's remove old events except the last one: we need it to calculate some curves.
+  CleanupEventsOlderThan(aTime);
+
+  for (size_t bufferIndex = 0; bufferIndex < aSize; ++bufferIndex, ++aTime) {
+
+    bool timeMatchesEventIndex = false;
+    const AudioTimelineEvent* next;
+    for (; ; ++eventIndex) {
+
+      if (eventIndex >= mEvents.Length()) {
+        next = nullptr;
+        break;
+      }
+
+      next = &mEvents[eventIndex];
+      if (aTime < TimeOf(*next)) {
+        break;
+      }
+
+#ifdef DEBUG
+      MOZ_ASSERT(next->mType == AudioTimelineEvent::SetValueAtTime ||
+                 next->mType == AudioTimelineEvent::SetTarget ||
+                 next->mType == AudioTimelineEvent::LinearRamp ||
+                 next->mType == AudioTimelineEvent::ExponentialRamp ||
+                 next->mType == AudioTimelineEvent::SetValueCurve);
+#endif
+
+      if (TimesEqual(aTime, TimeOf(*next))) {
+        mLastComputedValue = mComputedValue;
+        // Find the last event with the same time
+        while (eventIndex < mEvents.Length() - 1 &&
+               TimesEqual(aTime, TimeOf(mEvents[eventIndex + 1]))) {
+          mLastComputedValue = GetValueAtTimeOfEvent<TimeType>(&mEvents[eventIndex]);
+          ++eventIndex;
+        }
+
+        timeMatchesEventIndex = true;
+        break;
+      }
+
+      previous = next;
+    }
+
+    if (timeMatchesEventIndex) {
+      // The time matches one of the events exactly.
+      MOZ_ASSERT(TimesEqual(aTime, TimeOf(mEvents[eventIndex])));
+      mComputedValue = GetValueAtTimeOfEvent<TimeType>(&mEvents[eventIndex]);
+    } else {
+      mComputedValue = GetValuesAtTimeHelperInternal(aTime, previous, next);
+    }
+
+    aBuffer[bufferIndex] = mComputedValue;
+  }
+}
+template void
+AudioEventTimeline::GetValuesAtTimeHelper(double aTime, float* aBuffer,
+                                          const size_t aSize);
+template void
+AudioEventTimeline::GetValuesAtTimeHelper(int64_t aTime, float* aBuffer,
+                                          const size_t aSize);
+
+template<class TimeType> float
+AudioEventTimeline::GetValueAtTimeOfEvent(const AudioTimelineEvent* aNext)
+{
+  TimeType time = aNext->template Time<TimeType>();
+  switch (aNext->mType) {
+    case AudioTimelineEvent::SetTarget:
+      // SetTarget nodes can be handled no matter what their next node is
+      // (if they have one).
+      // Follow the curve, without regard to the next event, starting at
+      // the last value of the last event.
+      return ExponentialApproach(time,
+                                 mLastComputedValue, aNext->mValue,
+                                 aNext->mTimeConstant, time);
+      break;
+    case AudioTimelineEvent::SetValueCurve:
+      // SetValueCurve events can be handled no matter what their event
+      // node is (if they have one)
+      return ExtractValueFromCurve(time,
+                                   aNext->mCurve,
+                                   aNext->mCurveLength,
+                                   aNext->mDuration, time);
+      break;
+    default:
+      // For other event types
+      return aNext->mValue;
+  }
+}
+
+template<class TimeType> float
+AudioEventTimeline::GetValuesAtTimeHelperInternal(TimeType aTime,
+                                    const AudioTimelineEvent* aPrevious,
+                                    const AudioTimelineEvent* aNext)
+{
+  // If the requested time is before all of the existing events
+  if (!aPrevious) {
+     return mValue;
+  }
+
+  auto TimeOf = [](const AudioTimelineEvent* aEvent) -> TimeType {
+    return aEvent->template Time<TimeType>();
+  };
+
+  // SetTarget nodes can be handled no matter what their next node is (if
+  // they have one)
+  if (aPrevious->mType == AudioTimelineEvent::SetTarget) {
+    return ExponentialApproach(TimeOf(aPrevious),
+                               mLastComputedValue, aPrevious->mValue,
+                               aPrevious->mTimeConstant, aTime);
+  }
+
+  // SetValueCurve events can be handled no matter what their next node is
+  // (if they have one)
+  if (aPrevious->mType == AudioTimelineEvent::SetValueCurve) {
+    return ExtractValueFromCurve(TimeOf(aPrevious),
+                                 aPrevious->mCurve, aPrevious->mCurveLength,
+                                 aPrevious->mDuration, aTime);
+  }
+
+  // If the requested time is after all of the existing events
+  if (!aNext) {
+    switch (aPrevious->mType) {
+      case AudioTimelineEvent::SetValueAtTime:
+      case AudioTimelineEvent::LinearRamp:
+      case AudioTimelineEvent::ExponentialRamp:
+        // The value will be constant after the last event
+        return aPrevious->mValue;
+      case AudioTimelineEvent::SetValueCurve:
+        return ExtractValueFromCurve(TimeOf(aPrevious),
+                                     aPrevious->mCurve, aPrevious->mCurveLength,
+                                     aPrevious->mDuration, aTime);
+      case AudioTimelineEvent::SetTarget:
+        MOZ_FALLTHROUGH_ASSERT("AudioTimelineEvent::SetTarget");
+      case AudioTimelineEvent::SetValue:
+      case AudioTimelineEvent::Cancel:
+      case AudioTimelineEvent::Stream:
+        MOZ_ASSERT(false, "Should have been handled earlier.");
+    }
+    MOZ_ASSERT(false, "unreached");
+  }
+
+  // Finally, handle the case where we have both a previous and a next event
+
+  // First, handle the case where our range ends up in a ramp event
+  switch (aNext->mType) {
+  case AudioTimelineEvent::LinearRamp:
+    return LinearInterpolate(TimeOf(aPrevious),
+                             aPrevious->mValue,
+                             TimeOf(aNext),
+                             aNext->mValue, aTime);
+
+  case AudioTimelineEvent::ExponentialRamp:
+    return ExponentialInterpolate(TimeOf(aPrevious),
+                                  aPrevious->mValue,
+                                  TimeOf(aNext),
+                                  aNext->mValue, aTime);
+
+  case AudioTimelineEvent::SetValueAtTime:
+  case AudioTimelineEvent::SetTarget:
+  case AudioTimelineEvent::SetValueCurve:
+    break;
+  case AudioTimelineEvent::SetValue:
+  case AudioTimelineEvent::Cancel:
+  case AudioTimelineEvent::Stream:
+    MOZ_ASSERT(false, "Should have been handled earlier.");
+  }
+
+  // Now handle all other cases
+  switch (aPrevious->mType) {
+  case AudioTimelineEvent::SetValueAtTime:
+  case AudioTimelineEvent::LinearRamp:
+  case AudioTimelineEvent::ExponentialRamp:
+    // If the next event type is neither linear or exponential ramp, the
+    // value is constant.
+    return aPrevious->mValue;
+  case AudioTimelineEvent::SetValueCurve:
+    return ExtractValueFromCurve(TimeOf(aPrevious),
+                                 aPrevious->mCurve, aPrevious->mCurveLength,
+                                 aPrevious->mDuration, aTime);
+  case AudioTimelineEvent::SetTarget:
+    MOZ_FALLTHROUGH_ASSERT("AudioTimelineEvent::SetTarget");
+  case AudioTimelineEvent::SetValue:
+  case AudioTimelineEvent::Cancel:
+  case AudioTimelineEvent::Stream:
+    MOZ_ASSERT(false, "Should have been handled earlier.");
+  }
+
+  MOZ_ASSERT(false, "unreached");
+  return 0.0f;
+}
+template float
+AudioEventTimeline::GetValuesAtTimeHelperInternal(double aTime,
+                                    const AudioTimelineEvent* aPrevious,
+                                    const AudioTimelineEvent* aNext);
+template float
+AudioEventTimeline::GetValuesAtTimeHelperInternal(int64_t aTime,
+                                    const AudioTimelineEvent* aPrevious,
+                                    const AudioTimelineEvent* aNext);
+
+const AudioTimelineEvent*
+AudioEventTimeline::GetPreviousEvent(double aTime) const
+{
+  const AudioTimelineEvent* previous = nullptr;
+  const AudioTimelineEvent* next = nullptr;
+
+  auto TimeOf = [](const AudioTimelineEvent& aEvent) -> double {
+    return aEvent.template Time<double>();
+  };
+
+  bool bailOut = false;
+  for (unsigned i = 0; !bailOut && i < mEvents.Length(); ++i) {
+    switch (mEvents[i].mType) {
+    case AudioTimelineEvent::SetValueAtTime:
+    case AudioTimelineEvent::SetTarget:
+    case AudioTimelineEvent::LinearRamp:
+    case AudioTimelineEvent::ExponentialRamp:
+    case AudioTimelineEvent::SetValueCurve:
+      if (aTime == TimeOf(mEvents[i])) {
+        // Find the last event with the same time
+        do {
+          ++i;
+        } while (i < mEvents.Length() &&
+                 aTime == TimeOf(mEvents[i]));
+        return &mEvents[i - 1];
+      }
+      previous = next;
+      next = &mEvents[i];
+      if (aTime < TimeOf(mEvents[i])) {
+        bailOut = true;
+      }
+      break;
+    default:
+      MOZ_ASSERT(false, "unreached");
+    }
+  }
+  // Handle the case where the time is past all of the events
+  if (!bailOut) {
+    previous = next;
+  }
+
+  return previous;
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/AudioEventTimeline.h b/dom/media/webaudio/AudioEventTimeline.h
new file mode 100644
index 0000000000..ae06ad4db9
--- /dev/null
+++ b/dom/media/webaudio/AudioEventTimeline.h
@@ -0,0 +1,474 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioEventTimeline_h_
+#define AudioEventTimeline_h_
+
+#include <algorithm>
+#include "mozilla/Assertions.h"
+#include "mozilla/FloatingPoint.h"
+#include "mozilla/PodOperations.h"
+
+#include "MainThreadUtils.h"
+#include "nsTArray.h"
+#include "math.h"
+#include "WebAudioUtils.h"
+
+namespace mozilla {
+
+class MediaStream;
+
+namespace dom {
+
+struct AudioTimelineEvent final
+{
+  enum Type : uint32_t
+  {
+    SetValue,
+    SetValueAtTime,
+    LinearRamp,
+    ExponentialRamp,
+    SetTarget,
+    SetValueCurve,
+    Stream,
+    Cancel
+  };
+
+  AudioTimelineEvent(Type aType, double aTime, float aValue, double aTimeConstant = 0.0,
+                     double aDuration = 0.0, const float* aCurve = nullptr,
+                     uint32_t aCurveLength = 0)
+    : mType(aType)
+    , mCurve(nullptr)
+    , mTimeConstant(aTimeConstant)
+    , mDuration(aDuration)
+#ifdef DEBUG
+    , mTimeIsInTicks(false)
+#endif
+  {
+    mTime = aTime;
+    if (aType == AudioTimelineEvent::SetValueCurve) {
+      SetCurveParams(aCurve, aCurveLength);
+    } else {
+      mValue = aValue;
+    }
+  }
+
+  explicit AudioTimelineEvent(MediaStream* aStream)
+    : mType(Stream)
+    , mCurve(nullptr)
+    , mStream(aStream)
+    , mTimeConstant(0.0)
+    , mDuration(0.0)
+#ifdef DEBUG
+    , mTimeIsInTicks(false)
+#endif
+  {
+  }
+
+  AudioTimelineEvent(const AudioTimelineEvent& rhs)
+  {
+    PodCopy(this, &rhs, 1);
+
+    if (rhs.mType == AudioTimelineEvent::SetValueCurve) {
+      SetCurveParams(rhs.mCurve, rhs.mCurveLength);
+    } else if (rhs.mType == AudioTimelineEvent::Stream) {
+      new (&mStream) decltype(mStream)(rhs.mStream);
+    }
+  }
+
+  ~AudioTimelineEvent()
+  {
+    if (mType == AudioTimelineEvent::SetValueCurve) {
+      delete[] mCurve;
+    }
+  }
+
+  template <class TimeType>
+  TimeType Time() const;
+
+  void SetTimeInTicks(int64_t aTimeInTicks)
+  {
+    mTimeInTicks = aTimeInTicks;
+#ifdef DEBUG
+    mTimeIsInTicks = true;
+#endif
+  }
+
+  void SetCurveParams(const float* aCurve, uint32_t aCurveLength) {
+    mCurveLength = aCurveLength;
+    if (aCurveLength) {
+      mCurve = new float[aCurveLength];
+      PodCopy(mCurve, aCurve, aCurveLength);
+    } else {
+      mCurve = nullptr;
+    }
+  }
+
+  Type mType;
+  union {
+    float mValue;
+    uint32_t mCurveLength;
+  };
+  // mCurve contains a buffer of SetValueCurve samples.  We sample the
+  // values in the buffer depending on how far along we are in time.
+  // If we're at time T and the event has started as time T0 and has a
+  // duration of D, we sample the buffer at floor(mCurveLength*(T-T0)/D)
+  // if T<T0+D, and just take the last sample in the buffer otherwise.
+  float* mCurve;
+  RefPtr<MediaStream> mStream;
+  double mTimeConstant;
+  double mDuration;
+#ifdef DEBUG
+  bool mTimeIsInTicks;
+#endif
+
+private:
+  // This member is accessed using the `Time` method, for safety.
+  //
+  // The time for an event can either be in absolute value or in ticks.
+  // Initially the time of the event is always in absolute value.
+  // In order to convert it to ticks, call SetTimeInTicks.  Once this
+  // method has been called for an event, the time cannot be converted
+  // back to absolute value.
+  union {
+    double mTime;
+    int64_t mTimeInTicks;
+  };
+};
+
+template <>
+inline double AudioTimelineEvent::Time<double>() const
+{
+  MOZ_ASSERT(!mTimeIsInTicks);
+  return mTime;
+}
+
+template <>
+inline int64_t AudioTimelineEvent::Time<int64_t>() const
+{
+  MOZ_ASSERT(!NS_IsMainThread());
+  MOZ_ASSERT(mTimeIsInTicks);
+  return mTimeInTicks;
+}
+
+/**
+ * Some methods in this class will be instantiated with different ErrorResult
+ * template arguments for testing and production code.
+ *
+ * ErrorResult is a type which satisfies the following:
+ *  - Implements a Throw() method taking an nsresult argument, representing an error code.
+ */
+class AudioEventTimeline
+{
+public:
+  explicit AudioEventTimeline(float aDefaultValue)
+    : mValue(aDefaultValue),
+      mComputedValue(aDefaultValue),
+      mLastComputedValue(aDefaultValue)
+  { }
+
+  template <class ErrorResult>
+  bool ValidateEvent(AudioTimelineEvent& aEvent, ErrorResult& aRv)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    auto TimeOf = [](const AudioTimelineEvent& aEvent) -> double {
+      return aEvent.template Time<double>();
+    };
+
+    // Validate the event itself
+    if (!WebAudioUtils::IsTimeValid(TimeOf(aEvent)) ||
+        !WebAudioUtils::IsTimeValid(aEvent.mTimeConstant)) {
+      aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+      return false;
+    }
+
+    if (aEvent.mType == AudioTimelineEvent::SetValueCurve) {
+      if (!aEvent.mCurve || !aEvent.mCurveLength) {
+        aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+        return false;
+      }
+      for (uint32_t i = 0; i < aEvent.mCurveLength; ++i) {
+        if (!IsValid(aEvent.mCurve[i])) {
+          aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+          return false;
+        }
+      }
+    }
+
+    bool timeAndValueValid = IsValid(aEvent.mValue) &&
+                             IsValid(aEvent.mDuration);
+    if (!timeAndValueValid) {
+      aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+      return false;
+    }
+
+    // Make sure that non-curve events don't fall within the duration of a
+    // curve event.
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (mEvents[i].mType == AudioTimelineEvent::SetValueCurve &&
+          !(aEvent.mType == AudioTimelineEvent::SetValueCurve &&
+            TimeOf(aEvent) == TimeOf(mEvents[i])) &&
+          TimeOf(mEvents[i]) <= TimeOf(aEvent) &&
+          TimeOf(mEvents[i]) + mEvents[i].mDuration >= TimeOf(aEvent)) {
+        aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+        return false;
+      }
+    }
+
+    // Make sure that curve events don't fall in a range which includes other
+    // events.
+    if (aEvent.mType == AudioTimelineEvent::SetValueCurve) {
+      for (unsigned i = 0; i < mEvents.Length(); ++i) {
+        // In case we have two curve at the same time
+        if (mEvents[i].mType == AudioTimelineEvent::SetValueCurve &&
+            TimeOf(mEvents[i]) == TimeOf(aEvent)) {
+          continue;
+        }
+        if (TimeOf(mEvents[i]) > TimeOf(aEvent) &&
+            TimeOf(mEvents[i]) < TimeOf(aEvent) + aEvent.mDuration) {
+          aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+          return false;
+        }
+      }
+    }
+
+    // Make sure that invalid values are not used for exponential curves
+    if (aEvent.mType == AudioTimelineEvent::ExponentialRamp) {
+      if (aEvent.mValue <= 0.f) {
+        aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+        return false;
+      }
+      const AudioTimelineEvent* previousEvent = GetPreviousEvent(TimeOf(aEvent));
+      if (previousEvent) {
+        if (previousEvent->mValue <= 0.f) {
+          aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+          return false;
+        }
+      } else {
+        if (mValue <= 0.f) {
+          aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+          return false;
+        }
+      }
+    }
+    return true;
+  }
+
+  template<typename TimeType>
+  void InsertEvent(const AudioTimelineEvent& aEvent)
+  {
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (aEvent.template Time<TimeType>() == mEvents[i].template Time<TimeType>()) {
+        if (aEvent.mType == mEvents[i].mType) {
+          // If times and types are equal, replace the event
+          mEvents.ReplaceElementAt(i, aEvent);
+        } else {
+          // Otherwise, place the element after the last event of another type
+          do {
+            ++i;
+          } while (i < mEvents.Length() &&
+                   aEvent.mType != mEvents[i].mType &&
+                   aEvent.template Time<TimeType>() == mEvents[i].template Time<TimeType>());
+          mEvents.InsertElementAt(i, aEvent);
+        }
+        return;
+      }
+      // Otherwise, place the event right after the latest existing event
+      if (aEvent.template Time<TimeType>() < mEvents[i].template Time<TimeType>()) {
+        mEvents.InsertElementAt(i, aEvent);
+        return;
+      }
+    }
+
+    // If we couldn't find a place for the event, just append it to the list
+    mEvents.AppendElement(aEvent);
+  }
+
+  bool HasSimpleValue() const
+  {
+    return mEvents.IsEmpty();
+  }
+
+  float GetValue() const
+  {
+    // This method should only be called if HasSimpleValue() returns true
+    MOZ_ASSERT(HasSimpleValue());
+    return mValue;
+  }
+
+  float Value() const
+  {
+    // TODO: Return the current value based on the timeline of the AudioContext
+    return mValue;
+  }
+
+  void SetValue(float aValue)
+  {
+    // Silently don't change anything if there are any events
+    if (mEvents.IsEmpty()) {
+      mLastComputedValue = mComputedValue = mValue = aValue;
+    }
+  }
+
+  template <class ErrorResult>
+  void SetValueAtTime(float aValue, double aStartTime, ErrorResult& aRv)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::SetValueAtTime, aStartTime, aValue);
+
+    if (ValidateEvent(event, aRv)) {
+      InsertEvent<double>(event);
+    }
+  }
+
+  template <class ErrorResult>
+  void LinearRampToValueAtTime(float aValue, double aEndTime, ErrorResult& aRv)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::LinearRamp, aEndTime, aValue);
+
+    if (ValidateEvent(event, aRv)) {
+      InsertEvent<double>(event);
+    }
+  }
+
+  template <class ErrorResult>
+  void ExponentialRampToValueAtTime(float aValue, double aEndTime, ErrorResult& aRv)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::ExponentialRamp, aEndTime, aValue);
+
+    if (ValidateEvent(event, aRv)) {
+      InsertEvent<double>(event);
+    }
+  }
+
+  template <class ErrorResult>
+  void SetTargetAtTime(float aTarget, double aStartTime, double aTimeConstant, ErrorResult& aRv)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::SetTarget, aStartTime, aTarget, aTimeConstant);
+
+    if (ValidateEvent(event, aRv)) {
+      InsertEvent<double>(event);
+    }
+  }
+
+  template <class ErrorResult>
+  void SetValueCurveAtTime(const float* aValues, uint32_t aValuesLength, double aStartTime, double aDuration, ErrorResult& aRv)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::SetValueCurve, aStartTime, 0.0f, 0.0f, aDuration, aValues, aValuesLength);
+    if (ValidateEvent(event, aRv)) {
+      InsertEvent<double>(event);
+    }
+  }
+
+  template<typename TimeType>
+  void CancelScheduledValues(TimeType aStartTime)
+  {
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (mEvents[i].template Time<TimeType>() >= aStartTime) {
+#ifdef DEBUG
+        // Sanity check: the array should be sorted, so all of the following
+        // events should have a time greater than aStartTime too.
+        for (unsigned j = i + 1; j < mEvents.Length(); ++j) {
+          MOZ_ASSERT(mEvents[j].template Time<TimeType>() >= aStartTime);
+        }
+#endif
+        mEvents.TruncateLength(i);
+        break;
+      }
+    }
+  }
+
+  void CancelAllEvents()
+  {
+    mEvents.Clear();
+  }
+
+  static bool TimesEqual(int64_t aLhs, int64_t aRhs)
+  {
+    return aLhs == aRhs;
+  }
+
+  // Since we are going to accumulate error by adding 0.01 multiple time in a
+  // loop, we want to fuzz the equality check in GetValueAtTime.
+  static bool TimesEqual(double aLhs, double aRhs)
+  {
+    const float kEpsilon = 0.0000000001f;
+    return fabs(aLhs - aRhs) < kEpsilon;
+  }
+
+  template<class TimeType>
+  float GetValueAtTime(TimeType aTime)
+  {
+    float result;
+    GetValuesAtTimeHelper(aTime, &result, 1);
+    return result;
+  }
+
+  template<class TimeType>
+  void GetValuesAtTime(TimeType aTime, float* aBuffer, const size_t aSize)
+  {
+    MOZ_ASSERT(aBuffer);
+    GetValuesAtTimeHelper(aTime, aBuffer, aSize);
+  }
+
+  // Return the number of events scheduled
+  uint32_t GetEventCount() const
+  {
+    return mEvents.Length();
+  }
+
+  template<class TimeType>
+  void CleanupEventsOlderThan(TimeType aTime)
+  {
+    while (mEvents.Length() > 1 &&
+        aTime > mEvents[1].template Time<TimeType>()) {
+
+      if (mEvents[1].mType == AudioTimelineEvent::SetTarget) {
+        mLastComputedValue = GetValuesAtTimeHelperInternal(
+                                mEvents[1].template Time<TimeType>(),
+                                &mEvents[0], nullptr);
+      }
+
+      mEvents.RemoveElementAt(0);
+    }
+  }
+
+private:
+  template<class TimeType>
+  void GetValuesAtTimeHelper(TimeType aTime, float* aBuffer, const size_t aSize);
+
+  template<class TimeType>
+  float GetValueAtTimeOfEvent(const AudioTimelineEvent* aNext);
+
+  template<class TimeType>
+  float GetValuesAtTimeHelperInternal(TimeType aTime,
+                                      const AudioTimelineEvent* aPrevious,
+                                      const AudioTimelineEvent* aNext);
+
+  const AudioTimelineEvent* GetPreviousEvent(double aTime) const;
+
+  static bool IsValid(double value)
+  {
+    return mozilla::IsFinite(value);
+  }
+
+  // This is a sorted array of the events in the timeline.  Queries of this
+  // data structure should probably be more frequent than modifications to it,
+  // and that is the reason why we're using a simple array as the data structure.
+  // We can optimize this in the future if the performance of the array ends up
+  // being a bottleneck.
+  nsTArray<AudioTimelineEvent> mEvents;
+  float mValue;
+  // This is the value of this AudioParam we computed at the last tick.
+  float mComputedValue;
+  // This is the value of this AudioParam at the last tick of the previous event.
+  float mLastComputedValue;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioListener.cpp b/dom/media/webaudio/AudioListener.cpp
new file mode 100644
index 0000000000..0bd11156a4
--- /dev/null
+++ b/dom/media/webaudio/AudioListener.cpp
@@ -0,0 +1,131 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioListener.h"
+#include "AudioContext.h"
+#include "mozilla/dom/AudioListenerBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(AudioListener, mContext)
+
+NS_IMPL_CYCLE_COLLECTION_ROOT_NATIVE(AudioListener, AddRef)
+NS_IMPL_CYCLE_COLLECTION_UNROOT_NATIVE(AudioListener, Release)
+
+AudioListener::AudioListener(AudioContext* aContext)
+  : mContext(aContext)
+  , mPosition()
+  , mFrontVector(0., 0., -1.)
+  , mRightVector(1., 0., 0.)
+  , mVelocity()
+  , mDopplerFactor(1.)
+  , mSpeedOfSound(343.3) // meters/second
+{
+  MOZ_ASSERT(aContext);
+}
+
+JSObject*
+AudioListener::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioListenerBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+AudioListener::SetOrientation(double aX, double aY, double aZ,
+                              double aXUp, double aYUp, double aZUp)
+{
+  ThreeDPoint front(aX, aY, aZ);
+  // The panning effect and the azimuth and elevation calculation in the Web
+  // Audio spec becomes undefined with linearly dependent vectors, so keep
+  // existing state in these situations.
+  if (front.IsZero()) {
+    return;
+  }
+  // Normalize before using CrossProduct() to avoid overflow.
+  front.Normalize();
+  ThreeDPoint up(aXUp, aYUp, aZUp);
+  if (up.IsZero()) {
+    return;
+  }
+  up.Normalize();
+  ThreeDPoint right = front.CrossProduct(up);
+  if (right.IsZero()) {
+    return;
+  }
+  right.Normalize();
+
+  if (!mFrontVector.FuzzyEqual(front)) {
+    mFrontVector = front;
+    SendThreeDPointParameterToStream(PannerNode::LISTENER_FRONT_VECTOR, front);
+  }
+  if (!mRightVector.FuzzyEqual(right)) {
+    mRightVector = right;
+    SendThreeDPointParameterToStream(PannerNode::LISTENER_RIGHT_VECTOR, right);
+  }
+}
+
+void
+AudioListener::RegisterPannerNode(PannerNode* aPannerNode)
+{
+  mPanners.AppendElement(aPannerNode);
+
+  // Let the panner node know about our parameters
+  aPannerNode->SendThreeDPointParameterToStream(PannerNode::LISTENER_POSITION, mPosition);
+  aPannerNode->SendThreeDPointParameterToStream(PannerNode::LISTENER_FRONT_VECTOR, mFrontVector);
+  aPannerNode->SendThreeDPointParameterToStream(PannerNode::LISTENER_RIGHT_VECTOR, mRightVector);
+  aPannerNode->SendThreeDPointParameterToStream(PannerNode::LISTENER_VELOCITY, mVelocity);
+  aPannerNode->SendDoubleParameterToStream(PannerNode::LISTENER_DOPPLER_FACTOR, mDopplerFactor);
+  aPannerNode->SendDoubleParameterToStream(PannerNode::LISTENER_SPEED_OF_SOUND, mSpeedOfSound);
+  UpdatePannersVelocity();
+}
+
+void AudioListener::UnregisterPannerNode(PannerNode* aPannerNode)
+{
+  mPanners.RemoveElement(aPannerNode);
+}
+
+void
+AudioListener::SendDoubleParameterToStream(uint32_t aIndex, double aValue)
+{
+  for (uint32_t i = 0; i < mPanners.Length(); ++i) {
+    if (mPanners[i]) {
+      mPanners[i]->SendDoubleParameterToStream(aIndex, aValue);
+    }
+  }
+}
+
+void
+AudioListener::SendThreeDPointParameterToStream(uint32_t aIndex, const ThreeDPoint& aValue)
+{
+  for (uint32_t i = 0; i < mPanners.Length(); ++i) {
+    if (mPanners[i]) {
+      mPanners[i]->SendThreeDPointParameterToStream(aIndex, aValue);
+    }
+  }
+}
+
+void AudioListener::UpdatePannersVelocity()
+{
+  for (uint32_t i = 0; i < mPanners.Length(); ++i) {
+    if (mPanners[i]) {
+      mPanners[i]->SendDopplerToSourcesIfNeeded();
+    }
+  }
+}
+
+size_t
+AudioListener::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = aMallocSizeOf(this);
+  // AudioNodes are tracked separately
+  amount += mPanners.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/AudioListener.h b/dom/media/webaudio/AudioListener.h
new file mode 100644
index 0000000000..e3eaf1ca42
--- /dev/null
+++ b/dom/media/webaudio/AudioListener.h
@@ -0,0 +1,133 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioListener_h_
+#define AudioListener_h_
+
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "ThreeDPoint.h"
+#include "AudioContext.h"
+#include "PannerNode.h"
+#include "WebAudioUtils.h"
+#include "js/TypeDecls.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+
+namespace dom {
+
+class AudioListener final : public nsWrapperCache
+{
+public:
+  explicit AudioListener(AudioContext* aContext);
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioListener)
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(AudioListener)
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  AudioContext* GetParentObject() const
+  {
+    return mContext;
+  }
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  double DopplerFactor() const
+  {
+    return mDopplerFactor;
+  }
+  void SetDopplerFactor(double aDopplerFactor)
+  {
+    if (WebAudioUtils::FuzzyEqual(mDopplerFactor, aDopplerFactor)) {
+      return;
+    }
+    mDopplerFactor = aDopplerFactor;
+    SendDoubleParameterToStream(PannerNode::LISTENER_DOPPLER_FACTOR, mDopplerFactor);
+  }
+
+  double SpeedOfSound() const
+  {
+    return mSpeedOfSound;
+  }
+  void SetSpeedOfSound(double aSpeedOfSound)
+  {
+    if (WebAudioUtils::FuzzyEqual(mSpeedOfSound, aSpeedOfSound)) {
+      return;
+    }
+    mSpeedOfSound = aSpeedOfSound;
+    SendDoubleParameterToStream(PannerNode::LISTENER_SPEED_OF_SOUND, mSpeedOfSound);
+  }
+
+  void SetPosition(double aX, double aY, double aZ)
+  {
+    if (WebAudioUtils::FuzzyEqual(mPosition.x, aX) &&
+        WebAudioUtils::FuzzyEqual(mPosition.y, aY) &&
+        WebAudioUtils::FuzzyEqual(mPosition.z, aZ)) {
+      return;
+    }
+    mPosition.x = aX;
+    mPosition.y = aY;
+    mPosition.z = aZ;
+    SendThreeDPointParameterToStream(PannerNode::LISTENER_POSITION, mPosition);
+  }
+
+  const ThreeDPoint& Position() const
+  {
+    return mPosition;
+  }
+
+  void SetOrientation(double aX, double aY, double aZ,
+                      double aXUp, double aYUp, double aZUp);
+
+  const ThreeDPoint& Velocity() const
+  {
+    return mVelocity;
+  }
+
+  void SetVelocity(double aX, double aY, double aZ)
+  {
+    if (WebAudioUtils::FuzzyEqual(mVelocity.x, aX) &&
+        WebAudioUtils::FuzzyEqual(mVelocity.y, aY) &&
+        WebAudioUtils::FuzzyEqual(mVelocity.z, aZ)) {
+      return;
+    }
+    mVelocity.x = aX;
+    mVelocity.y = aY;
+    mVelocity.z = aZ;
+    SendThreeDPointParameterToStream(PannerNode::LISTENER_VELOCITY, mVelocity);
+    UpdatePannersVelocity();
+  }
+
+  void RegisterPannerNode(PannerNode* aPannerNode);
+  void UnregisterPannerNode(PannerNode* aPannerNode);
+
+private:
+  ~AudioListener() {}
+
+  void SendDoubleParameterToStream(uint32_t aIndex, double aValue);
+  void SendThreeDPointParameterToStream(uint32_t aIndex, const ThreeDPoint& aValue);
+  void UpdatePannersVelocity();
+
+private:
+  friend class PannerNode;
+  RefPtr<AudioContext> mContext;
+  ThreeDPoint mPosition;
+  ThreeDPoint mFrontVector;
+  ThreeDPoint mRightVector;
+  ThreeDPoint mVelocity;
+  double mDopplerFactor;
+  double mSpeedOfSound;
+  nsTArray<WeakPtr<PannerNode> > mPanners;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioNode.cpp b/dom/media/webaudio/AudioNode.cpp
new file mode 100644
index 0000000000..2b64fcf880
--- /dev/null
+++ b/dom/media/webaudio/AudioNode.cpp
@@ -0,0 +1,666 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNode.h"
+#include "mozilla/ErrorResult.h"
+#include "AudioNodeStream.h"
+#include "AudioNodeEngine.h"
+#include "mozilla/dom/AudioParam.h"
+#include "mozilla/Services.h"
+#include "nsIObserverService.h"
+
+namespace mozilla {
+namespace dom {
+
+static const uint32_t INVALID_PORT = 0xffffffff;
+static uint32_t gId = 0;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(AudioNode, DOMEventTargetHelper)
+  tmp->DisconnectFromGraph();
+  if (tmp->mContext) {
+    tmp->mContext->UnregisterNode(tmp);
+  }
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mContext)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mOutputNodes)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mOutputParams)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioNode,
+                                                  DOMEventTargetHelper)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mContext)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOutputNodes)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOutputParams)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_ADDREF_INHERITED(AudioNode, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(AudioNode, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioNode)
+  NS_INTERFACE_MAP_ENTRY(nsISupportsWeakReference)
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+AudioNode::AudioNode(AudioContext* aContext,
+                     uint32_t aChannelCount,
+                     ChannelCountMode aChannelCountMode,
+                     ChannelInterpretation aChannelInterpretation)
+  : DOMEventTargetHelper(aContext->GetParentObject())
+  , mContext(aContext)
+  , mChannelCount(aChannelCount)
+  , mChannelCountMode(aChannelCountMode)
+  , mChannelInterpretation(aChannelInterpretation)
+  , mId(gId++)
+  , mPassThrough(false)
+{
+  MOZ_ASSERT(aContext);
+  DOMEventTargetHelper::BindToOwner(aContext->GetParentObject());
+  aContext->RegisterNode(this);
+}
+
+AudioNode::~AudioNode()
+{
+  MOZ_ASSERT(mInputNodes.IsEmpty());
+  MOZ_ASSERT(mOutputNodes.IsEmpty());
+  MOZ_ASSERT(mOutputParams.IsEmpty());
+  MOZ_ASSERT(!mStream,
+             "The webaudio-node-demise notification must have been sent");
+  if (mContext) {
+    mContext->UnregisterNode(this);
+  }
+}
+
+size_t
+AudioNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  // Not owned:
+  // - mContext
+  // - mStream
+  size_t amount = 0;
+
+  amount += mInputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mInputNodes.Length(); i++) {
+    amount += mInputNodes[i].SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  // Just measure the array. The entire audio node graph is measured via the
+  // MediaStreamGraph's streams, so we don't want to double-count the elements.
+  amount += mOutputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+
+  amount += mOutputParams.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mOutputParams.Length(); i++) {
+    amount += mOutputParams[i]->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t
+AudioNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+template <class InputNode>
+static size_t
+FindIndexOfNode(const nsTArray<InputNode>& aInputNodes, const AudioNode* aNode)
+{
+  for (size_t i = 0; i < aInputNodes.Length(); ++i) {
+    if (aInputNodes[i].mInputNode == aNode) {
+      return i;
+    }
+  }
+  return nsTArray<InputNode>::NoIndex;
+}
+
+template <class InputNode>
+static size_t
+FindIndexOfNodeWithPorts(const nsTArray<InputNode>& aInputNodes,
+                         const AudioNode* aNode,
+                         uint32_t aInputPort, uint32_t aOutputPort)
+{
+  for (size_t i = 0; i < aInputNodes.Length(); ++i) {
+    if (aInputNodes[i].mInputNode == aNode &&
+        aInputNodes[i].mInputPort == aInputPort &&
+        aInputNodes[i].mOutputPort == aOutputPort) {
+      return i;
+    }
+  }
+  return nsTArray<InputNode>::NoIndex;
+}
+
+void
+AudioNode::DisconnectFromGraph()
+{
+  MOZ_ASSERT(mRefCnt.get() > mInputNodes.Length(),
+             "Caller should be holding a reference");
+
+  // The idea here is that we remove connections one by one, and at each step
+  // the graph is in a valid state.
+
+  // Disconnect inputs. We don't need them anymore.
+  while (!mInputNodes.IsEmpty()) {
+    size_t i = mInputNodes.Length() - 1;
+    RefPtr<AudioNode> input = mInputNodes[i].mInputNode;
+    mInputNodes.RemoveElementAt(i);
+    input->mOutputNodes.RemoveElement(this);
+  }
+
+  while (!mOutputNodes.IsEmpty()) {
+    size_t i = mOutputNodes.Length() - 1;
+    RefPtr<AudioNode> output = mOutputNodes[i].forget();
+    mOutputNodes.RemoveElementAt(i);
+    size_t inputIndex = FindIndexOfNode(output->mInputNodes, this);
+    // It doesn't matter which one we remove, since we're going to remove all
+    // entries for this node anyway.
+    output->mInputNodes.RemoveElementAt(inputIndex);
+    // This effects of this connection will remain.
+    output->NotifyHasPhantomInput();
+  }
+
+  while (!mOutputParams.IsEmpty()) {
+    size_t i = mOutputParams.Length() - 1;
+    RefPtr<AudioParam> output = mOutputParams[i].forget();
+    mOutputParams.RemoveElementAt(i);
+    size_t inputIndex = FindIndexOfNode(output->InputNodes(), this);
+    // It doesn't matter which one we remove, since we're going to remove all
+    // entries for this node anyway.
+    output->RemoveInputNode(inputIndex);
+  }
+
+  DestroyMediaStream();
+}
+
+AudioNode*
+AudioNode::Connect(AudioNode& aDestination, uint32_t aOutput,
+                   uint32_t aInput, ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs() ||
+      aInput >= aDestination.NumberOfInputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return nullptr;
+  }
+
+  if (Context() != aDestination.Context()) {
+    aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+    return nullptr;
+  }
+
+  if (FindIndexOfNodeWithPorts(aDestination.mInputNodes,
+                               this, aInput, aOutput) !=
+      nsTArray<AudioNode::InputNode>::NoIndex) {
+    // connection already exists.
+    return &aDestination;
+  }
+
+  WEB_AUDIO_API_LOG("%f: %s %u Connect() to %s %u",
+                    Context()->CurrentTime(), NodeType(), Id(),
+                    aDestination.NodeType(), aDestination.Id());
+
+  // The MediaStreamGraph will handle cycle detection. We don't need to do it
+  // here.
+
+  mOutputNodes.AppendElement(&aDestination);
+  InputNode* input = aDestination.mInputNodes.AppendElement();
+  input->mInputNode = this;
+  input->mInputPort = aInput;
+  input->mOutputPort = aOutput;
+  AudioNodeStream* destinationStream = aDestination.mStream;
+  if (mStream && destinationStream) {
+    // Connect streams in the MediaStreamGraph
+    MOZ_ASSERT(aInput <= UINT16_MAX, "Unexpected large input port number");
+    MOZ_ASSERT(aOutput <= UINT16_MAX, "Unexpected large output port number");
+    input->mStreamPort = destinationStream->
+      AllocateInputPort(mStream, AudioNodeStream::AUDIO_TRACK, TRACK_ANY,
+                        static_cast<uint16_t>(aInput),
+                        static_cast<uint16_t>(aOutput));
+  }
+  aDestination.NotifyInputsChanged();
+
+  // This connection may have connected a panner and a source.
+  Context()->UpdatePannerSource();
+
+  return &aDestination;
+}
+
+void
+AudioNode::Connect(AudioParam& aDestination, uint32_t aOutput,
+                   ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  if (Context() != aDestination.GetParentObject()) {
+    aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+    return;
+  }
+
+  if (FindIndexOfNodeWithPorts(aDestination.InputNodes(),
+                               this, INVALID_PORT, aOutput) !=
+      nsTArray<AudioNode::InputNode>::NoIndex) {
+    // connection already exists.
+    return;
+  }
+
+  mOutputParams.AppendElement(&aDestination);
+  InputNode* input = aDestination.AppendInputNode();
+  input->mInputNode = this;
+  input->mInputPort = INVALID_PORT;
+  input->mOutputPort = aOutput;
+
+  MediaStream* stream = aDestination.Stream();
+  MOZ_ASSERT(stream->AsProcessedStream());
+  ProcessedMediaStream* ps = static_cast<ProcessedMediaStream*>(stream);
+  if (mStream) {
+    // Setup our stream as an input to the AudioParam's stream
+    MOZ_ASSERT(aOutput <= UINT16_MAX, "Unexpected large output port number");
+    input->mStreamPort =
+      ps->AllocateInputPort(mStream, AudioNodeStream::AUDIO_TRACK, TRACK_ANY,
+                            0, static_cast<uint16_t>(aOutput));
+  }
+}
+
+void
+AudioNode::SendDoubleParameterToStream(uint32_t aIndex, double aValue)
+{
+  MOZ_ASSERT(mStream, "How come we don't have a stream here?");
+  mStream->SetDoubleParameter(aIndex, aValue);
+}
+
+void
+AudioNode::SendInt32ParameterToStream(uint32_t aIndex, int32_t aValue)
+{
+  MOZ_ASSERT(mStream, "How come we don't have a stream here?");
+  mStream->SetInt32Parameter(aIndex, aValue);
+}
+
+void
+AudioNode::SendThreeDPointParameterToStream(uint32_t aIndex,
+                                            const ThreeDPoint& aValue)
+{
+  MOZ_ASSERT(mStream, "How come we don't have a stream here?");
+  mStream->SetThreeDPointParameter(aIndex, aValue);
+}
+
+void
+AudioNode::SendChannelMixingParametersToStream()
+{
+  if (mStream) {
+    mStream->SetChannelMixingParameters(mChannelCount, mChannelCountMode,
+                                        mChannelInterpretation);
+  }
+}
+
+template<>
+bool
+AudioNode::DisconnectFromOutputIfConnected<AudioNode>(uint32_t aOutputNodeIndex,
+                                                      uint32_t aInputIndex)
+{
+  WEB_AUDIO_API_LOG("%f: %s %u Disconnect()", Context()->CurrentTime(),
+                    NodeType(), Id());
+
+  AudioNode* destination = mOutputNodes[aOutputNodeIndex];
+
+  MOZ_ASSERT(aOutputNodeIndex < mOutputNodes.Length());
+  MOZ_ASSERT(aInputIndex < destination->InputNodes().Length());
+
+  // An upstream node may be starting to play on the graph thread, and the
+  // engine for a downstream node may be sending a PlayingRefChangeHandler
+  // ADDREF message to this (main) thread.  Wait for a round trip before
+  // releasing nodes, to give engines receiving sound now time to keep their
+  // nodes alive.
+  class RunnableRelease final : public Runnable
+  {
+  public:
+    explicit RunnableRelease(already_AddRefed<AudioNode> aNode)
+      : mNode(aNode) {}
+
+    NS_IMETHOD Run() override
+    {
+      mNode = nullptr;
+      return NS_OK;
+    }
+  private:
+    RefPtr<AudioNode> mNode;
+  };
+
+  InputNode& input = destination->mInputNodes[aInputIndex];
+  if (input.mInputNode != this) {
+    return false;
+  }
+
+  // Remove one instance of 'dest' from mOutputNodes. There could be
+  // others, and it's not correct to remove them all since some of them
+  // could be for different output ports.
+  RefPtr<AudioNode> output = mOutputNodes[aOutputNodeIndex].forget();
+  mOutputNodes.RemoveElementAt(aOutputNodeIndex);
+  // Destroying the InputNode here sends a message to the graph thread
+  // to disconnect the streams, which should be sent before the
+  // RunAfterPendingUpdates() call below.
+  destination->mInputNodes.RemoveElementAt(aInputIndex);
+  output->NotifyInputsChanged();
+  if (mStream) {
+    nsCOMPtr<nsIRunnable> runnable = new RunnableRelease(output.forget());
+    mStream->RunAfterPendingUpdates(runnable.forget());
+  }
+  return true;
+}
+
+template<>
+bool
+AudioNode::DisconnectFromOutputIfConnected<AudioParam>(uint32_t aOutputParamIndex,
+                                                       uint32_t aInputIndex)
+{
+  MOZ_ASSERT(aOutputParamIndex < mOutputParams.Length());
+
+  AudioParam* destination = mOutputParams[aOutputParamIndex];
+
+  MOZ_ASSERT(aInputIndex < destination->InputNodes().Length());
+
+  const InputNode& input = destination->InputNodes()[aInputIndex];
+  if (input.mInputNode != this) {
+    return false;
+  }
+  destination->RemoveInputNode(aInputIndex);
+  // Remove one instance of 'dest' from mOutputParams. There could be
+  // others, and it's not correct to remove them all since some of them
+  // could be for different output ports.
+  mOutputParams.RemoveElementAt(aOutputParamIndex);
+  return true;
+}
+
+template<>
+const nsTArray<AudioNode::InputNode>&
+AudioNode::InputsForDestination<AudioNode>(uint32_t aOutputNodeIndex) const {
+  return mOutputNodes[aOutputNodeIndex]->InputNodes();
+}
+
+template<>
+const nsTArray<AudioNode::InputNode>&
+AudioNode::InputsForDestination<AudioParam>(uint32_t aOutputNodeIndex) const {
+  return mOutputParams[aOutputNodeIndex]->InputNodes();
+}
+
+template<typename DestinationType, typename Predicate>
+bool
+AudioNode::DisconnectMatchingDestinationInputs(uint32_t aDestinationIndex,
+                                               Predicate aPredicate)
+{
+  bool wasConnected = false;
+  uint32_t inputCount =
+    InputsForDestination<DestinationType>(aDestinationIndex).Length();
+
+  for (int32_t inputIndex = inputCount - 1; inputIndex >= 0; --inputIndex) {
+    const InputNode& input =
+      InputsForDestination<DestinationType>(aDestinationIndex)[inputIndex];
+    if (aPredicate(input)) {
+      if (DisconnectFromOutputIfConnected<DestinationType>(aDestinationIndex,
+                                                           inputIndex)) {
+        wasConnected = true;
+        break;
+      }
+    }
+  }
+  return wasConnected;
+}
+
+void
+AudioNode::Disconnect(ErrorResult& aRv)
+{
+  for (int32_t outputIndex = mOutputNodes.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioNode>(outputIndex,
+                                                   [](const InputNode&) {
+                                                     return true;
+                                                   });
+  }
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioParam>(outputIndex,
+                                                    [](const InputNode&) {
+                                                      return true;
+                                                    });
+  }
+
+  // This disconnection may have disconnected a panner and a source.
+  Context()->UpdatePannerSource();
+}
+
+void
+AudioNode::Disconnect(uint32_t aOutput, ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex,
+        [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioParam>(
+        outputIndex,
+        [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+
+  // This disconnection may have disconnected a panner and a source.
+  Context()->UpdatePannerSource();
+}
+
+void
+AudioNode::Disconnect(AudioNode& aDestination, ErrorResult& aRv)
+{
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |=
+      DisconnectMatchingDestinationInputs<AudioNode>(outputIndex,
+                                                     [](const InputNode&) {
+                                                       return true;
+                                                     });
+  }
+
+  if (!wasConnected) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_ACCESS_ERR);
+    return;
+  }
+
+  // This disconnection may have disconnected a panner and a source.
+  Context()->UpdatePannerSource();
+}
+
+void
+AudioNode::Disconnect(AudioNode& aDestination,
+                      uint32_t aOutput,
+                      ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |=
+      DisconnectMatchingDestinationInputs<AudioNode>(
+          outputIndex,
+          [aOutput](const InputNode& aInputNode) {
+            return aInputNode.mOutputPort == aOutput;
+          });
+  }
+
+  if (!wasConnected) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_ACCESS_ERR);
+    return;
+  }
+
+  // This disconnection may have disconnected a panner and a source.
+  Context()->UpdatePannerSource();
+}
+
+void
+AudioNode::Disconnect(AudioNode& aDestination,
+                      uint32_t aOutput,
+                      uint32_t aInput,
+                      ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  if (aInput >= aDestination.NumberOfInputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |=
+      DisconnectMatchingDestinationInputs<AudioNode>(
+          outputIndex,
+          [aOutput, aInput](const InputNode& aInputNode) {
+            return aInputNode.mOutputPort == aOutput &&
+                   aInputNode.mInputPort == aInput;
+          });
+  }
+
+  if (!wasConnected) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_ACCESS_ERR);
+    return;
+  }
+
+  // This disconnection may have disconnected a panner and a source.
+  Context()->UpdatePannerSource();
+}
+
+void
+AudioNode::Disconnect(AudioParam& aDestination, ErrorResult& aRv)
+{
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    if (mOutputParams[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |=
+      DisconnectMatchingDestinationInputs<AudioParam>(outputIndex,
+                                                      [](const InputNode&) {
+                                                        return true;
+                                                      });
+  }
+
+  if (!wasConnected) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_ACCESS_ERR);
+    return;
+  }
+}
+
+void
+AudioNode::Disconnect(AudioParam& aDestination,
+                      uint32_t aOutput,
+                      ErrorResult& aRv)
+{
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1;
+       outputIndex >= 0; --outputIndex) {
+    if (mOutputParams[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |=
+      DisconnectMatchingDestinationInputs<AudioParam>(
+          outputIndex,
+          [aOutput](const InputNode& aInputNode) {
+            return aInputNode.mOutputPort == aOutput;
+          });
+  }
+
+  if (!wasConnected) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_ACCESS_ERR);
+    return;
+  }
+}
+
+void
+AudioNode::DestroyMediaStream()
+{
+  if (mStream) {
+    // Remove the node pointer on the engine.
+    AudioNodeStream* ns = mStream;
+    MOZ_ASSERT(ns, "How come we don't have a stream here?");
+    MOZ_ASSERT(ns->Engine()->NodeMainThread() == this,
+               "Invalid node reference");
+    ns->Engine()->ClearNode();
+
+    mStream->Destroy();
+    mStream = nullptr;
+
+    nsCOMPtr<nsIObserverService> obs = services::GetObserverService();
+    if (obs) {
+      nsAutoString id;
+      id.AppendPrintf("%u", mId);
+      obs->NotifyObservers(nullptr, "webaudio-node-demise", id.get());
+    }
+  }
+}
+
+void
+AudioNode::RemoveOutputParam(AudioParam* aParam)
+{
+  mOutputParams.RemoveElement(aParam);
+}
+
+bool
+AudioNode::PassThrough() const
+{
+  MOZ_ASSERT(NumberOfInputs() <= 1 && NumberOfOutputs() == 1);
+  return mPassThrough;
+}
+
+void
+AudioNode::SetPassThrough(bool aPassThrough)
+{
+  MOZ_ASSERT(NumberOfInputs() <= 1 && NumberOfOutputs() == 1);
+  mPassThrough = aPassThrough;
+  if (mStream) {
+    mStream->SetPassThrough(mPassThrough);
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioNode.h b/dom/media/webaudio/AudioNode.h
new file mode 100644
index 0000000000..ebef129c83
--- /dev/null
+++ b/dom/media/webaudio/AudioNode.h
@@ -0,0 +1,294 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioNode_h_
+#define AudioNode_h_
+
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/dom/AudioNodeBinding.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsTArray.h"
+#include "AudioContext.h"
+#include "MediaStreamGraph.h"
+#include "WebAudioUtils.h"
+#include "mozilla/MemoryReporting.h"
+#include "nsWeakReference.h"
+#include "SelfRef.h"
+
+namespace mozilla {
+
+namespace dom {
+
+class AudioContext;
+class AudioBufferSourceNode;
+class AudioParam;
+class AudioParamTimeline;
+struct ThreeDPoint;
+
+/**
+ * The DOM object representing a Web Audio AudioNode.
+ *
+ * Each AudioNode has a MediaStream representing the actual
+ * real-time processing and output of this AudioNode.
+ *
+ * We track the incoming and outgoing connections to other AudioNodes.
+ * Outgoing connections have strong ownership.  Also, AudioNodes that will
+ * produce sound on their output even when they have silent or no input ask
+ * the AudioContext to keep playing or tail-time references to keep them alive
+ * until the context is finished.
+ *
+ * Explicit disconnections will only remove references from output nodes after
+ * the graph is notified and the main thread receives a reply.  Similarly,
+ * nodes with playing or tail-time references release these references only
+ * after receiving notification from their engine on the graph thread that
+ * playing has stopped.  Engines notifying the main thread that they have
+ * finished do so strictly *after* producing and returning their last block.
+ * In this way, an engine that receives non-null input knows that the input
+ * comes from nodes that are still alive and will keep their output nodes
+ * alive for at least as long as it takes to process messages from the graph
+ * thread.  i.e. the engine receiving non-null input knows that its node is
+ * still alive, and will still be alive when it receives a message from the
+ * engine.
+ */
+class AudioNode : public DOMEventTargetHelper,
+                  public nsSupportsWeakReference
+{
+protected:
+  // You can only use refcounting to delete this object
+  virtual ~AudioNode();
+
+public:
+  AudioNode(AudioContext* aContext,
+            uint32_t aChannelCount,
+            ChannelCountMode aChannelCountMode,
+            ChannelInterpretation aChannelInterpretation);
+
+  // This should be idempotent (safe to call multiple times).
+  virtual void DestroyMediaStream();
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioNode,
+                                           DOMEventTargetHelper)
+
+  virtual AudioBufferSourceNode* AsAudioBufferSourceNode()
+  {
+    return nullptr;
+  }
+
+  AudioContext* GetParentObject() const
+  {
+    return mContext;
+  }
+
+  AudioContext* Context() const
+  {
+    return mContext;
+  }
+
+  virtual AudioNode* Connect(AudioNode& aDestination, uint32_t aOutput,
+                             uint32_t aInput, ErrorResult& aRv);
+
+  virtual void Connect(AudioParam& aDestination, uint32_t aOutput,
+                       ErrorResult& aRv);
+
+  virtual void Disconnect(ErrorResult& aRv);
+  virtual void Disconnect(uint32_t aOutput, ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination, ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                          ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination,
+                          uint32_t aOutput, uint32_t aInput,
+                          ErrorResult& aRv);
+  virtual void Disconnect(AudioParam& aDestination, ErrorResult& aRv);
+  virtual void Disconnect(AudioParam& aDestination, uint32_t aOutput,
+                          ErrorResult& aRv);
+
+  // Called after input nodes have been explicitly added or removed through
+  // the Connect() or Disconnect() methods.
+  virtual void NotifyInputsChanged() {}
+  // Indicate that the node should continue indefinitely to behave as if an
+  // input is connected, even though there is no longer a corresponding entry
+  // in mInputNodes.  Called after an input node has been removed because it
+  // is being garbage collected.
+  virtual void NotifyHasPhantomInput() {}
+
+  // The following two virtual methods must be implemented by each node type
+  // to provide their number of input and output ports. These numbers are
+  // constant for the lifetime of the node. Both default to 1.
+  virtual uint16_t NumberOfInputs() const { return 1; }
+  virtual uint16_t NumberOfOutputs() const { return 1; }
+
+  uint32_t Id() const { return mId; }
+
+  bool PassThrough() const;
+  void SetPassThrough(bool aPassThrough);
+
+  uint32_t ChannelCount() const { return mChannelCount; }
+  virtual void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv)
+  {
+    if (aChannelCount == 0 ||
+        aChannelCount > WebAudioUtils::MaxChannelCount) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    mChannelCount = aChannelCount;
+    SendChannelMixingParametersToStream();
+  }
+  ChannelCountMode ChannelCountModeValue() const
+  {
+    return mChannelCountMode;
+  }
+  virtual void SetChannelCountModeValue(ChannelCountMode aMode, ErrorResult& aRv)
+  {
+    mChannelCountMode = aMode;
+    SendChannelMixingParametersToStream();
+  }
+  ChannelInterpretation ChannelInterpretationValue() const
+  {
+    return mChannelInterpretation;
+  }
+  void SetChannelInterpretationValue(ChannelInterpretation aMode)
+  {
+    mChannelInterpretation = aMode;
+    SendChannelMixingParametersToStream();
+  }
+
+  struct InputNode final
+  {
+    ~InputNode()
+    {
+      if (mStreamPort) {
+        mStreamPort->Destroy();
+      }
+    }
+
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+    {
+      size_t amount = 0;
+      if (mStreamPort) {
+        amount += mStreamPort->SizeOfIncludingThis(aMallocSizeOf);
+      }
+
+      return amount;
+    }
+
+    // Weak reference.
+    AudioNode* mInputNode;
+    RefPtr<MediaInputPort> mStreamPort;
+    // The index of the input port this node feeds into.
+    // This is not used for connections to AudioParams.
+    uint32_t mInputPort;
+    // The index of the output port this node comes out of.
+    uint32_t mOutputPort;
+  };
+
+  // Returns the stream, if any.
+  AudioNodeStream* GetStream() const { return mStream; }
+
+  const nsTArray<InputNode>& InputNodes() const
+  {
+    return mInputNodes;
+  }
+  const nsTArray<RefPtr<AudioNode> >& OutputNodes() const
+  {
+    return mOutputNodes;
+  }
+  const nsTArray<RefPtr<AudioParam> >& OutputParams() const
+  {
+    return mOutputParams;
+  }
+
+  template<typename T>
+  const nsTArray<InputNode>&
+  InputsForDestination(uint32_t aOutputIndex) const;
+
+  void RemoveOutputParam(AudioParam* aParam);
+
+  // MarkActive() asks the context to keep the AudioNode alive until the
+  // context is finished.  This takes care of "playing" references and
+  // "tail-time" references.
+  void MarkActive() { Context()->RegisterActiveNode(this); }
+  // Active nodes call MarkInactive() when they have finished producing sound
+  // for the foreseeable future.
+  // Do not call MarkInactive from a node destructor.  If the destructor is
+  // called, then the node is already inactive.
+  // MarkInactive() may delete |this|.
+  void MarkInactive() { Context()->UnregisterActiveNode(this); }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const;
+
+  // Returns a string from constant static storage identifying the dom node
+  // type.
+  virtual const char* NodeType() const = 0;
+
+private:
+  // Given:
+  //
+  // - a DestinationType, that can be an AudioNode or an AudioParam ;
+  // - a Predicate, a function that takes an InputNode& and returns a bool ;
+  //
+  // This method iterates on the InputNodes() of the node at the index
+  // aDestinationIndex, and calls `DisconnectFromOutputIfConnected` with this
+  // input node, if aPredicate returns true.
+  template<typename DestinationType, typename Predicate>
+  bool DisconnectMatchingDestinationInputs(uint32_t aDestinationIndex,
+                                           Predicate aPredicate);
+
+  virtual void LastRelease() override
+  {
+    // We are about to be deleted, disconnect the object from the graph before
+    // the derived type is destroyed.
+    DisconnectFromGraph();
+  }
+  // Callers must hold a reference to 'this'.
+  void DisconnectFromGraph();
+
+  template<typename DestinationType>
+  bool DisconnectFromOutputIfConnected(uint32_t aOutputIndex, uint32_t aInputIndex);
+
+protected:
+  // Helpers for sending different value types to streams
+  void SendDoubleParameterToStream(uint32_t aIndex, double aValue);
+  void SendInt32ParameterToStream(uint32_t aIndex, int32_t aValue);
+  void SendThreeDPointParameterToStream(uint32_t aIndex, const ThreeDPoint& aValue);
+  void SendChannelMixingParametersToStream();
+
+private:
+  RefPtr<AudioContext> mContext;
+
+protected:
+  // Must be set in the constructor. Must not be null unless finished.
+  RefPtr<AudioNodeStream> mStream;
+
+private:
+  // For every InputNode, there is a corresponding entry in mOutputNodes of the
+  // InputNode's mInputNode.
+  nsTArray<InputNode> mInputNodes;
+  // For every mOutputNode entry, there is a corresponding entry in mInputNodes
+  // of the mOutputNode entry. We won't necessarily be able to identify the
+  // exact matching entry, since mOutputNodes doesn't include the port
+  // identifiers and the same node could be connected on multiple ports.
+  nsTArray<RefPtr<AudioNode> > mOutputNodes;
+  // For every mOutputParams entry, there is a corresponding entry in
+  // AudioParam::mInputNodes of the mOutputParams entry. We won't necessarily be
+  // able to identify the exact matching entry, since mOutputParams doesn't
+  // include the port identifiers and the same node could be connected on
+  // multiple ports.
+  nsTArray<RefPtr<AudioParam> > mOutputParams;
+  uint32_t mChannelCount;
+  ChannelCountMode mChannelCountMode;
+  ChannelInterpretation mChannelInterpretation;
+  const uint32_t mId;
+  // Whether the node just passes through its input.  This is a devtools API that
+  // only works for some node types.
+  bool mPassThrough;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioNodeEngine.cpp b/dom/media/webaudio/AudioNodeEngine.cpp
new file mode 100644
index 0000000000..91170adb31
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngine.cpp
@@ -0,0 +1,400 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngine.h"
+#ifdef BUILD_ARM_NEON
+#include "mozilla/arm.h"
+#include "AudioNodeEngineNEON.h"
+#endif
+#ifdef USE_SSE2
+#include "mozilla/SSE.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngineSSE2.h"
+#endif
+
+namespace mozilla {
+
+already_AddRefed<ThreadSharedFloatArrayBufferList>
+ThreadSharedFloatArrayBufferList::Create(uint32_t aChannelCount,
+                                         size_t aLength,
+                                         const mozilla::fallible_t&)
+{
+  RefPtr<ThreadSharedFloatArrayBufferList> buffer =
+    new ThreadSharedFloatArrayBufferList(aChannelCount);
+
+  for (uint32_t i = 0; i < aChannelCount; ++i) {
+    float* channelData = js_pod_malloc<float>(aLength);
+    if (!channelData) {
+      return nullptr;
+    }
+
+    buffer->SetData(i, channelData, js_free, channelData);
+  }
+
+  return buffer.forget();
+}
+
+void
+WriteZeroesToAudioBlock(AudioBlock* aChunk,
+                        uint32_t aStart, uint32_t aLength)
+{
+  MOZ_ASSERT(aStart + aLength <= WEBAUDIO_BLOCK_SIZE);
+  MOZ_ASSERT(!aChunk->IsNull(), "You should pass a non-null chunk");
+  if (aLength == 0)
+    return;
+
+  for (uint32_t i = 0; i < aChunk->ChannelCount(); ++i) {
+    PodZero(aChunk->ChannelFloatsForWrite(i) + aStart, aLength);
+  }
+}
+
+void AudioBufferCopyWithScale(const float* aInput,
+                              float aScale,
+                              float* aOutput,
+                              uint32_t aSize)
+{
+  if (aScale == 1.0f) {
+    PodCopy(aOutput, aInput, aSize);
+  } else {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] = aInput[i]*aScale;
+    }
+  }
+}
+
+void AudioBufferAddWithScale(const float* aInput,
+                             float aScale,
+                             float* aOutput,
+                             uint32_t aSize)
+{
+#ifdef BUILD_ARM_NEON
+  if (mozilla::supports_neon()) {
+    AudioBufferAddWithScale_NEON(aInput, aScale, aOutput, aSize);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    if (aScale == 1.0f) {
+      while (aSize && (!IS_ALIGNED16(aInput) || !IS_ALIGNED16(aOutput))) {
+        *aOutput += *aInput;
+        ++aOutput;
+        ++aInput;
+        --aSize;
+      }
+    } else {
+      while (aSize && (!IS_ALIGNED16(aInput) || !IS_ALIGNED16(aOutput))) {
+        *aOutput += *aInput*aScale;
+        ++aOutput;
+        ++aInput;
+        --aSize;
+      }
+    }
+
+    // we need to round aSize down to the nearest multiple of 16
+    uint32_t alignedSize = aSize & ~0x0F;
+    if (alignedSize > 0) {
+      AudioBufferAddWithScale_SSE(aInput, aScale, aOutput, alignedSize);
+
+      // adjust parameters for use with scalar operations below
+      aInput += alignedSize;
+      aOutput += alignedSize;
+      aSize -= alignedSize;
+    }
+  }
+#endif
+
+  if (aScale == 1.0f) {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] += aInput[i];
+    }
+  } else {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] += aInput[i]*aScale;
+    }
+  }
+}
+
+void
+AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                              float aScale,
+                              float aOutput[WEBAUDIO_BLOCK_SIZE])
+{
+  AudioBufferAddWithScale(aInput, aScale, aOutput, WEBAUDIO_BLOCK_SIZE);
+}
+
+void
+AudioBlockCopyChannelWithScale(const float* aInput,
+                               float aScale,
+                               float* aOutput)
+{
+  if (aScale == 1.0f) {
+    memcpy(aOutput, aInput, WEBAUDIO_BLOCK_SIZE*sizeof(float));
+  } else {
+#ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      AudioBlockCopyChannelWithScale_NEON(aInput, aScale, aOutput);
+      return;
+    }
+#endif
+
+#ifdef USE_SSE2
+    if (mozilla::supports_sse2()) {
+      AudioBlockCopyChannelWithScale_SSE(aInput, aScale, aOutput);
+      return;
+    }
+#endif
+
+    for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutput[i] = aInput[i]*aScale;
+    }
+  }
+}
+
+void
+BufferComplexMultiply(const float* aInput,
+                      const float* aScale,
+                      float* aOutput,
+                      uint32_t aSize)
+{
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse()) {
+    BufferComplexMultiply_SSE(aInput, aScale, aOutput, aSize);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < aSize * 2; i += 2) {
+    float real1 = aInput[i];
+    float imag1 = aInput[i + 1];
+    float real2 = aScale[i];
+    float imag2 = aScale[i + 1];
+    float realResult = real1 * real2 - imag1 * imag2;
+    float imagResult = real1 * imag2 + imag1 * real2;
+    aOutput[i] = realResult;
+    aOutput[i + 1] = imagResult;
+  }
+}
+
+float
+AudioBufferPeakValue(const float *aInput, uint32_t aSize)
+{
+  float max = 0.0f;
+  for (uint32_t i = 0; i < aSize; i++) {
+    float mag = fabs(aInput[i]);
+    if (mag > max) {
+      max = mag;
+    }
+  }
+  return max;
+}
+
+void
+AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                               const float aScale[WEBAUDIO_BLOCK_SIZE],
+                               float aOutput[WEBAUDIO_BLOCK_SIZE])
+{
+#ifdef BUILD_ARM_NEON
+  if (mozilla::supports_neon()) {
+    AudioBlockCopyChannelWithScale_NEON(aInput, aScale, aOutput);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    AudioBlockCopyChannelWithScale_SSE(aInput, aScale, aOutput);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    aOutput[i] = aInput[i]*aScale[i];
+  }
+}
+
+void
+AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
+                       float aScale)
+{
+  AudioBufferInPlaceScale(aBlock, aScale, WEBAUDIO_BLOCK_SIZE);
+}
+
+void
+AudioBufferInPlaceScale(float* aBlock,
+                        float aScale,
+                        uint32_t aSize)
+{
+  if (aScale == 1.0f) {
+    return;
+  }
+#ifdef BUILD_ARM_NEON
+  if (mozilla::supports_neon()) {
+    AudioBufferInPlaceScale_NEON(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    AudioBufferInPlaceScale_SSE(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    *aBlock++ *= aScale;
+  }
+}
+
+void
+AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                          float aGainL[WEBAUDIO_BLOCK_SIZE],
+                          float aGainR[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+  AudioBlockCopyChannelWithScale(aInput, aGainL, aOutputL);
+  AudioBlockCopyChannelWithScale(aInput, aGainR, aOutputR);
+}
+
+void
+AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                          float aGainL, float aGainR,
+                          float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+  AudioBlockCopyChannelWithScale(aInput, aGainL, aOutputL);
+  AudioBlockCopyChannelWithScale(aInput, aGainR, aOutputR);
+}
+
+void
+AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                            const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                            float aGainL, float aGainR, bool aIsOnTheLeft,
+                            float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+#ifdef BUILD_ARM_NEON
+  if (mozilla::supports_neon()) {
+    AudioBlockPanStereoToStereo_NEON(aInputL, aInputR,
+                                     aGainL, aGainR, aIsOnTheLeft,
+                                     aOutputL, aOutputR);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    AudioBlockPanStereoToStereo_SSE(aInputL, aInputR,
+                                    aGainL, aGainR, aIsOnTheLeft,
+                                    aOutputL, aOutputR);
+    return;
+  }
+#endif
+
+  uint32_t i;
+
+  if (aIsOnTheLeft) {
+    for (i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutputL[i] = aInputL[i] + aInputR[i] * aGainL;
+      aOutputR[i] = aInputR[i] * aGainR;
+    }
+  } else {
+    for (i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutputL[i] = aInputL[i] * aGainL;
+      aOutputR[i] = aInputR[i] + aInputL[i] * aGainR;
+    }
+  }
+}
+
+void
+AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                            const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                            float aGainL[WEBAUDIO_BLOCK_SIZE],
+                            float aGainR[WEBAUDIO_BLOCK_SIZE],
+                            bool  aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+#ifdef BUILD_ARM_NEON
+   if (mozilla::supports_neon()) {
+     AudioBlockPanStereoToStereo_NEON(aInputL, aInputR,
+                                      aGainL, aGainR, aIsOnTheLeft,
+                                      aOutputL, aOutputR);
+    return;
+  }
+#endif
+
+  uint32_t i;
+  for (i = 0; i < WEBAUDIO_BLOCK_SIZE; i++) {
+    if (aIsOnTheLeft[i]) {
+      aOutputL[i] = aInputL[i] + aInputR[i] * aGainL[i];
+      aOutputR[i] = aInputR[i] * aGainR[i];
+    } else {
+      aOutputL[i] = aInputL[i] * aGainL[i];
+      aOutputR[i] = aInputR[i] + aInputL[i] * aGainR[i];
+    }
+  }
+}
+
+float
+AudioBufferSumOfSquares(const float* aInput, uint32_t aLength)
+{
+  float sum = 0.0f;
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse()) {
+    const float* alignedInput = ALIGNED16(aInput);
+    float vLength = (aLength >> 4) << 4;
+
+    // use scalar operations for any unaligned data at the beginning
+    while (aInput != alignedInput) {
+        sum += *aInput * *aInput;
+        ++aInput;
+    }
+
+    sum += AudioBufferSumOfSquares_SSE(alignedInput, vLength);
+
+    // adjust aInput and aLength to use scalar operations for any
+    // remaining values
+    aInput = alignedInput + 1;
+    aLength -= vLength;
+  }
+#endif
+
+  while (aLength--) {
+    sum += *aInput * *aInput;
+    ++aInput;
+  }
+  return sum;
+}
+
+void
+AudioNodeEngine::ProcessBlock(AudioNodeStream* aStream,
+                              GraphTime aFrom,
+                              const AudioBlock& aInput,
+                              AudioBlock* aOutput,
+                              bool* aFinished)
+{
+  MOZ_ASSERT(mInputCount <= 1 && mOutputCount <= 1);
+  *aOutput = aInput;
+}
+
+void
+AudioNodeEngine::ProcessBlocksOnPorts(AudioNodeStream* aStream,
+                                      const OutputChunks& aInput,
+                                      OutputChunks& aOutput,
+                                      bool* aFinished)
+{
+  MOZ_ASSERT(mInputCount > 1 || mOutputCount > 1);
+  // Only produce one output port, and drop all other input ports.
+  aOutput[0] = aInput[0];
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeEngine.h b/dom/media/webaudio/AudioNodeEngine.h
new file mode 100644
index 0000000000..d49b5c906f
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngine.h
@@ -0,0 +1,410 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef MOZILLA_AUDIONODEENGINE_H_
+#define MOZILLA_AUDIONODEENGINE_H_
+
+#include "AudioSegment.h"
+#include "mozilla/dom/AudioNode.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/Mutex.h"
+
+namespace mozilla {
+
+namespace dom {
+struct ThreeDPoint;
+class AudioParamTimeline;
+class DelayNodeEngine;
+struct AudioTimelineEvent;
+} // namespace dom
+
+class AudioBlock;
+class AudioNodeStream;
+
+/**
+ * This class holds onto a set of immutable channel buffers. The storage
+ * for the buffers must be malloced, but the buffer pointers and the malloc
+ * pointers can be different (e.g. if the buffers are contained inside
+ * some malloced object).
+ */
+class ThreadSharedFloatArrayBufferList final : public ThreadSharedObject
+{
+public:
+  /**
+   * Construct with null channel data pointers.
+   */
+  explicit ThreadSharedFloatArrayBufferList(uint32_t aCount)
+  {
+    mContents.SetLength(aCount);
+  }
+  /**
+   * Create with buffers suitable for transfer to
+   * JS_NewArrayBufferWithContents().  The buffer contents are uninitialized
+   * and so should be set using GetDataForWrite().
+   */
+  static already_AddRefed<ThreadSharedFloatArrayBufferList>
+  Create(uint32_t aChannelCount, size_t aLength, const mozilla::fallible_t&);
+
+  struct Storage final
+  {
+    Storage() :
+      mDataToFree(nullptr),
+      mFree(nullptr),
+      mSampleData(nullptr)
+    {}
+    ~Storage() {
+      if (mFree) {
+        mFree(mDataToFree);
+      } else { MOZ_ASSERT(!mDataToFree); }
+    }
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+    {
+      // NB: mSampleData might not be owned, if it is it just points to
+      //     mDataToFree.
+      return aMallocSizeOf(mDataToFree);
+    }
+    void* mDataToFree;
+    void (*mFree)(void*);
+    float* mSampleData;
+  };
+
+  /**
+   * This can be called on any thread.
+   */
+  uint32_t GetChannels() const { return mContents.Length(); }
+  /**
+   * This can be called on any thread.
+   */
+  const float* GetData(uint32_t aIndex) const { return mContents[aIndex].mSampleData; }
+  /**
+   * This can be called on any thread, but only when the calling thread is the
+   * only owner.
+   */
+  float* GetDataForWrite(uint32_t aIndex)
+  {
+    MOZ_ASSERT(!IsShared());
+    return mContents[aIndex].mSampleData;
+  }
+
+  /**
+   * Call this only during initialization, before the object is handed to
+   * any other thread.
+   */
+  void SetData(uint32_t aIndex, void* aDataToFree, void (*aFreeFunc)(void*), float* aData)
+  {
+    Storage* s = &mContents[aIndex];
+    if (s->mFree) {
+      s->mFree(s->mDataToFree);
+    } else {
+      MOZ_ASSERT(!s->mDataToFree);
+    }
+
+    s->mDataToFree = aDataToFree;
+    s->mFree = aFreeFunc;
+    s->mSampleData = aData;
+  }
+
+  /**
+   * Put this object into an error state where there are no channels.
+   */
+  void Clear() { mContents.Clear(); }
+
+  size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = ThreadSharedObject::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mContents.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < mContents.Length(); i++) {
+      amount += mContents[i].SizeOfExcludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  AutoTArray<Storage, 2> mContents;
+};
+
+/**
+ * aChunk must have been allocated by AllocateAudioBlock.
+ */
+void WriteZeroesToAudioBlock(AudioBlock* aChunk, uint32_t aStart,
+                             uint32_t aLength);
+
+/**
+ * Copy with scale. aScale == 1.0f should be optimized.
+ */
+void AudioBufferCopyWithScale(const float* aInput,
+                              float aScale,
+                              float* aOutput,
+                              uint32_t aSize);
+
+/**
+ * Pointwise multiply-add operation. aScale == 1.0f should be optimized.
+ */
+void AudioBufferAddWithScale(const float* aInput,
+                             float aScale,
+                             float* aOutput,
+                             uint32_t aSize);
+
+/**
+ * Pointwise multiply-add operation. aScale == 1.0f should be optimized.
+ */
+void AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                   float aScale,
+                                   float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Pointwise copy-scaled operation. aScale == 1.0f should be optimized.
+ *
+ * Buffer size is implicitly assumed to be WEBAUDIO_BLOCK_SIZE.
+ */
+void AudioBlockCopyChannelWithScale(const float* aInput,
+                                    float aScale,
+                                    float* aOutput);
+
+/**
+ * Vector copy-scaled operation.
+ */
+void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                    const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                    float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Vector complex multiplication on arbitrary sized buffers.
+ */
+void BufferComplexMultiply(const float* aInput,
+                           const float* aScale,
+                           float* aOutput,
+                           uint32_t aSize);
+
+/**
+ * Vector maximum element magnitude ( max(abs(aInput)) ).
+ */
+float AudioBufferPeakValue(const float* aInput, uint32_t aSize);
+
+/**
+ * In place gain. aScale == 1.0f should be optimized.
+ */
+void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
+                            float aScale);
+
+/**
+ * In place gain. aScale == 1.0f should be optimized.
+ */
+void AudioBufferInPlaceScale(float* aBlock,
+                             float aScale,
+                             uint32_t aSize);
+
+/**
+ * Upmix a mono input to a stereo output, scaling the two output channels by two
+ * different gain value.
+ * This algorithm is specified in the WebAudio spec.
+ */
+void
+AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                          float aGainL, float aGainR,
+                          float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+void
+AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                          float aGainL[WEBAUDIO_BLOCK_SIZE],
+                          float aGainR[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                          float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+/**
+ * Pan a stereo source according to right and left gain, and the position
+ * (whether the listener is on the left of the source or not).
+ * This algorithm is specified in the WebAudio spec.
+ */
+void
+AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                            const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                            float aGainL, float aGainR, bool aIsOnTheLeft,
+                            float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+void
+AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                            const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                            float aGainL[WEBAUDIO_BLOCK_SIZE],
+                            float aGainR[WEBAUDIO_BLOCK_SIZE],
+                            bool  aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                            float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Return the sum of squares of all of the samples in the input.
+ */
+float
+AudioBufferSumOfSquares(const float* aInput, uint32_t aLength);
+
+/**
+ * All methods of this class and its subclasses are called on the
+ * MediaStreamGraph thread.
+ */
+class AudioNodeEngine
+{
+public:
+  // This should be compatible with AudioNodeStream::OutputChunks.
+  typedef AutoTArray<AudioBlock, 1> OutputChunks;
+
+  explicit AudioNodeEngine(dom::AudioNode* aNode)
+    : mNode(aNode)
+    , mNodeType(aNode ? aNode->NodeType() : nullptr)
+    , mInputCount(aNode ? aNode->NumberOfInputs() : 1)
+    , mOutputCount(aNode ? aNode->NumberOfOutputs() : 0)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    MOZ_COUNT_CTOR(AudioNodeEngine);
+  }
+  virtual ~AudioNodeEngine()
+  {
+    MOZ_ASSERT(!mNode, "The node reference must be already cleared");
+    MOZ_COUNT_DTOR(AudioNodeEngine);
+  }
+
+  virtual dom::DelayNodeEngine* AsDelayNodeEngine() { return nullptr; }
+
+  virtual void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam)
+  {
+    NS_ERROR("Invalid SetStreamTimeParameter index");
+  }
+  virtual void SetDoubleParameter(uint32_t aIndex, double aParam)
+  {
+    NS_ERROR("Invalid SetDoubleParameter index");
+  }
+  virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam)
+  {
+    NS_ERROR("Invalid SetInt32Parameter index");
+  }
+  virtual void RecvTimelineEvent(uint32_t aIndex,
+                                 dom::AudioTimelineEvent& aValue)
+  {
+    NS_ERROR("Invalid RecvTimelineEvent index");
+  }
+  virtual void SetThreeDPointParameter(uint32_t aIndex,
+                                       const dom::ThreeDPoint& aValue)
+  {
+    NS_ERROR("Invalid SetThreeDPointParameter index");
+  }
+  virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
+  {
+    NS_ERROR("SetBuffer called on engine that doesn't support it");
+  }
+  // This consumes the contents of aData.  aData will be emptied after this returns.
+  virtual void SetRawArrayData(nsTArray<float>& aData)
+  {
+    NS_ERROR("SetRawArrayData called on an engine that doesn't support it");
+  }
+
+  /**
+   * Produce the next block of audio samples, given input samples aInput
+   * (the mixed data for input 0).
+   * aInput is guaranteed to have float sample format (if it has samples at all)
+   * and to have been resampled to the sampling rate for the stream, and to have
+   * exactly WEBAUDIO_BLOCK_SIZE samples.
+   * *aFinished is set to false by the caller. The callee must not set this to
+   * true unless silent output is produced. If set to true, we'll finish the
+   * stream, consider this input inactive on any downstream nodes, and not
+   * call this again.
+   */
+  virtual void ProcessBlock(AudioNodeStream* aStream,
+                            GraphTime aFrom,
+                            const AudioBlock& aInput,
+                            AudioBlock* aOutput,
+                            bool* aFinished);
+  /**
+   * Produce the next block of audio samples, before input is provided.
+   * ProcessBlock() will be called later, and it then should not change
+   * aOutput.  This is used only for DelayNodeEngine in a feedback loop.
+   */
+  virtual void ProduceBlockBeforeInput(AudioNodeStream* aStream,
+                                       GraphTime aFrom,
+                                       AudioBlock* aOutput)
+  {
+    NS_NOTREACHED("ProduceBlockBeforeInput called on wrong engine\n");
+  }
+
+  /**
+   * Produce the next block of audio samples, given input samples in the aInput
+   * array.  There is one input sample per active port in aInput, in order.
+   * This is the multi-input/output version of ProcessBlock.  Only one kind
+   * of ProcessBlock is called on each node, depending on whether the
+   * number of inputs and outputs are both 1 or not.
+   *
+   * aInput is always guaranteed to not contain more input AudioChunks than the
+   * maximum number of inputs for the node.  It is the responsibility of the
+   * overrides of this function to make sure they will only add a maximum number
+   * of AudioChunks to aOutput as advertized by the AudioNode implementation.
+   * An engine may choose to produce fewer inputs than advertizes by the
+   * corresponding AudioNode, in which case it will be interpreted as a channel
+   * of silence.
+   */
+  virtual void ProcessBlocksOnPorts(AudioNodeStream* aStream,
+                                    const OutputChunks& aInput,
+                                    OutputChunks& aOutput,
+                                    bool* aFinished);
+
+  // IsActive() returns true if the engine needs to continue processing an
+  // unfinished stream even when it has silent or no input connections.  This
+  // includes tail-times and when sources have been scheduled to start.  If
+  // returning false, then the stream can be suspended.
+  virtual bool IsActive() const { return false; }
+
+  bool HasNode() const
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    return !!mNode;
+  }
+
+  dom::AudioNode* NodeMainThread() const
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    return mNode;
+  }
+
+  void ClearNode()
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    MOZ_ASSERT(mNode != nullptr);
+    mNode = nullptr;
+  }
+
+  uint16_t InputCount() const { return mInputCount; }
+  uint16_t OutputCount() const { return mOutputCount; }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    // NB: |mNode| is tracked separately so it is excluded here.
+    return 0;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  void SizeOfIncludingThis(MallocSizeOf aMallocSizeOf,
+                           AudioNodeSizes& aUsage) const
+  {
+    aUsage.mEngine = SizeOfIncludingThis(aMallocSizeOf);
+    aUsage.mNodeType = mNodeType;
+  }
+
+private:
+  dom::AudioNode* mNode; // main thread only
+  const char* const mNodeType;
+  const uint16_t mInputCount;
+  const uint16_t mOutputCount;
+};
+
+} // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODEENGINE_H_ */
diff --git a/dom/media/webaudio/AudioNodeEngineNEON.cpp b/dom/media/webaudio/AudioNodeEngineNEON.cpp
new file mode 100644
index 0000000000..079a1cc8bc
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineNEON.cpp
@@ -0,0 +1,318 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngineNEON.h"
+#include <arm_neon.h>
+
+//#ifdef DEBUG
+#if 0 // see bug 921099
+  #define ASSERT_ALIGNED(ptr)                                                  \
+            MOZ_ASSERT((((uintptr_t)ptr + 15) & ~0x0F) == (uintptr_t)ptr,      \
+                       #ptr " has to be aligned 16-bytes aligned.");
+#else
+  #define ASSERT_ALIGNED(ptr)
+#endif
+
+#define ADDRESS_OF(array, index) ((float32_t*)&array[index])
+
+namespace mozilla {
+void AudioBufferAddWithScale_NEON(const float* aInput,
+                                  float aScale,
+                                  float* aOutput,
+                                  uint32_t aSize)
+{
+  ASSERT_ALIGNED(aInput);
+  ASSERT_ALIGNED(aOutput);
+
+  float32x4_t vin0, vin1, vin2, vin3;
+  float32x4_t vout0, vout1, vout2, vout3;
+  float32x4_t vscale = vmovq_n_f32(aScale);
+
+  uint32_t dif = aSize % 16;
+  aSize -= dif;
+  unsigned i = 0;
+  for (; i < aSize; i+=16) {
+    vin0 = vld1q_f32(ADDRESS_OF(aInput, i));
+    vin1 = vld1q_f32(ADDRESS_OF(aInput, i+4));
+    vin2 = vld1q_f32(ADDRESS_OF(aInput, i+8));
+    vin3 = vld1q_f32(ADDRESS_OF(aInput, i+12));
+
+    vout0 = vld1q_f32(ADDRESS_OF(aOutput, i));
+    vout1 = vld1q_f32(ADDRESS_OF(aOutput, i+4));
+    vout2 = vld1q_f32(ADDRESS_OF(aOutput, i+8));
+    vout3 = vld1q_f32(ADDRESS_OF(aOutput, i+12));
+
+    vout0 = vmlaq_f32(vout0, vin0, vscale);
+    vout1 = vmlaq_f32(vout1, vin1, vscale);
+    vout2 = vmlaq_f32(vout2, vin2, vscale);
+    vout3 = vmlaq_f32(vout3, vin3, vscale);
+
+    vst1q_f32(ADDRESS_OF(aOutput, i), vout0);
+    vst1q_f32(ADDRESS_OF(aOutput, i+4), vout1);
+    vst1q_f32(ADDRESS_OF(aOutput, i+8), vout2);
+    vst1q_f32(ADDRESS_OF(aOutput, i+12), vout3);
+  }
+
+  for (unsigned j = 0; j < dif; ++i, ++j) {
+    aOutput[i] += aInput[i]*aScale;
+  }
+}
+void
+AudioBlockCopyChannelWithScale_NEON(const float* aInput,
+                                    float aScale,
+                                    float* aOutput)
+{
+  ASSERT_ALIGNED(aInput);
+  ASSERT_ALIGNED(aOutput);
+
+  float32x4_t vin0, vin1, vin2, vin3;
+  float32x4_t vout0, vout1, vout2, vout3;
+  float32x4_t vscale = vmovq_n_f32(aScale);
+
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=16) {
+    vin0 = vld1q_f32(ADDRESS_OF(aInput, i));
+    vin1 = vld1q_f32(ADDRESS_OF(aInput, i+4));
+    vin2 = vld1q_f32(ADDRESS_OF(aInput, i+8));
+    vin3 = vld1q_f32(ADDRESS_OF(aInput, i+12));
+
+    vout0 = vmulq_f32(vin0, vscale);
+    vout1 = vmulq_f32(vin1, vscale);
+    vout2 = vmulq_f32(vin2, vscale);
+    vout3 = vmulq_f32(vin3, vscale);
+
+    vst1q_f32(ADDRESS_OF(aOutput, i), vout0);
+    vst1q_f32(ADDRESS_OF(aOutput, i+4), vout1);
+    vst1q_f32(ADDRESS_OF(aOutput, i+8), vout2);
+    vst1q_f32(ADDRESS_OF(aOutput, i+12), vout3);
+  }
+}
+
+void
+AudioBlockCopyChannelWithScale_NEON(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                    const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                    float aOutput[WEBAUDIO_BLOCK_SIZE])
+{
+  ASSERT_ALIGNED(aInput);
+  ASSERT_ALIGNED(aScale);
+  ASSERT_ALIGNED(aOutput);
+
+  float32x4_t vin0, vin1, vin2, vin3;
+  float32x4_t vout0, vout1, vout2, vout3;
+  float32x4_t vscale0, vscale1, vscale2, vscale3;
+
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=16) {
+    vin0 = vld1q_f32(ADDRESS_OF(aInput, i));
+    vin1 = vld1q_f32(ADDRESS_OF(aInput, i+4));
+    vin2 = vld1q_f32(ADDRESS_OF(aInput, i+8));
+    vin3 = vld1q_f32(ADDRESS_OF(aInput, i+12));
+
+    vscale0 = vld1q_f32(ADDRESS_OF(aScale, i));
+    vscale1 = vld1q_f32(ADDRESS_OF(aScale, i+4));
+    vscale2 = vld1q_f32(ADDRESS_OF(aScale, i+8));
+    vscale3 = vld1q_f32(ADDRESS_OF(aScale, i+12));
+
+    vout0 = vmulq_f32(vin0, vscale0);
+    vout1 = vmulq_f32(vin1, vscale1);
+    vout2 = vmulq_f32(vin2, vscale2);
+    vout3 = vmulq_f32(vin3, vscale3);
+
+    vst1q_f32(ADDRESS_OF(aOutput, i), vout0);
+    vst1q_f32(ADDRESS_OF(aOutput, i+4), vout1);
+    vst1q_f32(ADDRESS_OF(aOutput, i+8), vout2);
+    vst1q_f32(ADDRESS_OF(aOutput, i+12), vout3);
+  }
+}
+
+void
+AudioBufferInPlaceScale_NEON(float* aBlock,
+                             float aScale,
+                             uint32_t aSize)
+{
+  ASSERT_ALIGNED(aBlock);
+
+  float32x4_t vin0, vin1, vin2, vin3;
+  float32x4_t vout0, vout1, vout2, vout3;
+  float32x4_t vscale = vmovq_n_f32(aScale);
+
+  uint32_t dif = aSize % 16;
+  uint32_t vectorSize = aSize - dif;
+  uint32_t i = 0;
+  for (; i < vectorSize; i+=16) {
+    vin0 = vld1q_f32(ADDRESS_OF(aBlock, i));
+    vin1 = vld1q_f32(ADDRESS_OF(aBlock, i+4));
+    vin2 = vld1q_f32(ADDRESS_OF(aBlock, i+8));
+    vin3 = vld1q_f32(ADDRESS_OF(aBlock, i+12));
+
+    vout0 = vmulq_f32(vin0, vscale);
+    vout1 = vmulq_f32(vin1, vscale);
+    vout2 = vmulq_f32(vin2, vscale);
+    vout3 = vmulq_f32(vin3, vscale);
+
+    vst1q_f32(ADDRESS_OF(aBlock, i), vout0);
+    vst1q_f32(ADDRESS_OF(aBlock, i+4), vout1);
+    vst1q_f32(ADDRESS_OF(aBlock, i+8), vout2);
+    vst1q_f32(ADDRESS_OF(aBlock, i+12), vout3);
+  }
+
+  for (unsigned j = 0; j < dif; ++i, ++j) {
+    aBlock[i] *= aScale;
+  }
+}
+
+void
+AudioBlockPanStereoToStereo_NEON(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL, float aGainR, bool aIsOnTheLeft,
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+  ASSERT_ALIGNED(aInputL);
+  ASSERT_ALIGNED(aInputR);
+  ASSERT_ALIGNED(aOutputL);
+  ASSERT_ALIGNED(aOutputR);
+
+  float32x4_t vinL0, vinL1;
+  float32x4_t vinR0, vinR1;
+  float32x4_t voutL0, voutL1;
+  float32x4_t voutR0, voutR1;
+  float32x4_t vscaleL = vmovq_n_f32(aGainL);
+  float32x4_t vscaleR = vmovq_n_f32(aGainR);
+
+  if (aIsOnTheLeft) {
+    for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=8) {
+      vinL0 = vld1q_f32(ADDRESS_OF(aInputL, i));
+      vinL1 = vld1q_f32(ADDRESS_OF(aInputL, i+4));
+
+      vinR0 = vld1q_f32(ADDRESS_OF(aInputR, i));
+      vinR1 = vld1q_f32(ADDRESS_OF(aInputR, i+4));
+
+      voutL0 = vmlaq_f32(vinL0, vinR0, vscaleL);
+      voutL1 = vmlaq_f32(vinL1, vinR1, vscaleL);
+
+      vst1q_f32(ADDRESS_OF(aOutputL, i), voutL0);
+      vst1q_f32(ADDRESS_OF(aOutputL, i+4), voutL1);
+
+      voutR0 = vmulq_f32(vinR0, vscaleR);
+      voutR1 = vmulq_f32(vinR1, vscaleR);
+
+      vst1q_f32(ADDRESS_OF(aOutputR, i), voutR0);
+      vst1q_f32(ADDRESS_OF(aOutputR, i+4), voutR1);
+    }
+  } else {
+    for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=8) {
+      vinL0 = vld1q_f32(ADDRESS_OF(aInputL, i));
+      vinL1 = vld1q_f32(ADDRESS_OF(aInputL, i+4));
+
+      vinR0 = vld1q_f32(ADDRESS_OF(aInputR, i));
+      vinR1 = vld1q_f32(ADDRESS_OF(aInputR, i+4));
+
+      voutL0 = vmulq_f32(vinL0, vscaleL);
+      voutL1 = vmulq_f32(vinL1, vscaleL);
+
+      vst1q_f32(ADDRESS_OF(aOutputL, i), voutL0);
+      vst1q_f32(ADDRESS_OF(aOutputL, i+4), voutL1);
+
+      voutR0 = vmlaq_f32(vinR0, vinL0, vscaleR);
+      voutR1 = vmlaq_f32(vinR1, vinL1, vscaleR);
+
+      vst1q_f32(ADDRESS_OF(aOutputR, i), voutR0);
+      vst1q_f32(ADDRESS_OF(aOutputR, i+4), voutR1);
+    }
+  }
+}
+
+void
+AudioBlockPanStereoToStereo_NEON(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainR[WEBAUDIO_BLOCK_SIZE],
+                                 const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+  ASSERT_ALIGNED(aInputL);
+  ASSERT_ALIGNED(aInputR);
+  ASSERT_ALIGNED(aGainL);
+  ASSERT_ALIGNED(aGainR);
+  ASSERT_ALIGNED(aIsOnTheLeft);
+  ASSERT_ALIGNED(aOutputL);
+  ASSERT_ALIGNED(aOutputR);
+
+  float32x4_t vinL0, vinL1;
+  float32x4_t vinR0, vinR1;
+  float32x4_t voutL0, voutL1;
+  float32x4_t voutR0, voutR1;
+  float32x4_t vscaleL0, vscaleL1;
+  float32x4_t vscaleR0, vscaleR1;
+  float32x4_t onleft0, onleft1, notonleft0, notonleft1;
+
+  float32x4_t zero = {0, 0, 0, 0};
+  uint8x8_t isOnTheLeft;
+
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=8) {
+    vinL0 = vld1q_f32(ADDRESS_OF(aInputL, i));
+    vinL1 = vld1q_f32(ADDRESS_OF(aInputL, i+4));
+
+    vinR0 = vld1q_f32(ADDRESS_OF(aInputR, i));
+    vinR1 = vld1q_f32(ADDRESS_OF(aInputR, i+4));
+
+    vscaleL0 = vld1q_f32(ADDRESS_OF(aGainL, i));
+    vscaleL1 = vld1q_f32(ADDRESS_OF(aGainL, i+4));
+
+    vscaleR0 = vld1q_f32(ADDRESS_OF(aGainR, i));
+    vscaleR1 = vld1q_f32(ADDRESS_OF(aGainR, i+4));
+
+    // Load output with boolean "on the left" values. This assumes that
+    // bools are stored as a single byte.
+    isOnTheLeft = vld1_u8((uint8_t *)&aIsOnTheLeft[i]);
+    voutL0 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 0), voutL0, 0);
+    voutL0 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 1), voutL0, 1);
+    voutL0 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 2), voutL0, 2);
+    voutL0 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 3), voutL0, 3);
+    voutL1 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 4), voutL1, 0);
+    voutL1 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 5), voutL1, 1);
+    voutL1 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 6), voutL1, 2);
+    voutL1 = vsetq_lane_f32(vget_lane_u8(isOnTheLeft, 7), voutL1, 3);
+
+    // Convert the boolean values into masks by setting all bits to 1
+    // if true.
+    voutL0 = (float32x4_t)vcgtq_f32(voutL0, zero);
+    voutL1 = (float32x4_t)vcgtq_f32(voutL1, zero);
+
+    // The right output masks are the same as the left masks
+    voutR0 = voutL0;
+    voutR1 = voutL1;
+
+    // Calculate left channel assuming isOnTheLeft
+    onleft0 = vmlaq_f32(vinL0, vinR0, vscaleL0);
+    onleft1 = vmlaq_f32(vinL1, vinR1, vscaleL0);
+
+    // Calculate left channel assuming not isOnTheLeft
+    notonleft0 = vmulq_f32(vinL0, vscaleL0);
+    notonleft1 = vmulq_f32(vinL1, vscaleL1);
+
+    // Write results using previously stored masks
+    voutL0 = vbslq_f32((uint32x4_t)voutL0, onleft0, notonleft0);
+    voutL1 = vbslq_f32((uint32x4_t)voutL1, onleft1, notonleft1);
+
+    // Calculate right channel assuming isOnTheLeft
+    onleft0 = vmulq_f32(vinR0, vscaleR0);
+    onleft1 = vmulq_f32(vinR1, vscaleR1);
+
+    // Calculate right channel assuming not isOnTheLeft
+    notonleft0 = vmlaq_f32(vinR0, vinL0, vscaleR0);
+    notonleft1 = vmlaq_f32(vinR1, vinL1, vscaleR1);
+
+    // Write results using previously stored masks
+    voutR0 = vbslq_f32((uint32x4_t)voutR0, onleft0, notonleft0);
+    voutR1 = vbslq_f32((uint32x4_t)voutR1, onleft1, notonleft1);
+
+    vst1q_f32(ADDRESS_OF(aOutputL, i), voutL0);
+    vst1q_f32(ADDRESS_OF(aOutputL, i+4), voutL1);
+    vst1q_f32(ADDRESS_OF(aOutputR, i), voutR0);
+    vst1q_f32(ADDRESS_OF(aOutputR, i+4), voutR1);
+  }
+}
+}
diff --git a/dom/media/webaudio/AudioNodeEngineNEON.h b/dom/media/webaudio/AudioNodeEngineNEON.h
new file mode 100644
index 0000000000..2b3e89b751
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineNEON.h
@@ -0,0 +1,49 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODEENGINENEON_H_
+#define MOZILLA_AUDIONODEENGINENEON_H_
+
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+void AudioBufferAddWithScale_NEON(const float* aInput,
+                                  float aScale,
+                                  float* aOutput,
+                                  uint32_t aSize);
+
+void
+AudioBlockCopyChannelWithScale_NEON(const float* aInput,
+                                    float aScale,
+                                    float* aOutput);
+
+void
+AudioBlockCopyChannelWithScale_NEON(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                    const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                    float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+void
+AudioBufferInPlaceScale_NEON(float* aBlock,
+                             float aScale,
+                             uint32_t aSize);
+
+void
+AudioBlockPanStereoToStereo_NEON(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL, float aGainR, bool aIsOnTheLeft,
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+void
+AudioBlockPanStereoToStereo_NEON(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainR[WEBAUDIO_BLOCK_SIZE],
+                                 const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+}
+
+#endif /* MOZILLA_AUDIONODEENGINENEON_H_ */
diff --git a/dom/media/webaudio/AudioNodeEngineSSE2.cpp b/dom/media/webaudio/AudioNodeEngineSSE2.cpp
new file mode 100644
index 0000000000..a033232390
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineSSE2.cpp
@@ -0,0 +1,315 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngineSSE2.h"
+#include "AlignmentUtils.h"
+#include <emmintrin.h>
+
+
+namespace mozilla {
+void
+AudioBufferAddWithScale_SSE(const float* aInput,
+                            float aScale,
+                            float* aOutput,
+                            uint32_t aSize)
+{
+  __m128 vin0, vin1, vin2, vin3,
+         vscaled0, vscaled1, vscaled2, vscaled3,
+         vout0, vout1, vout2, vout3,
+         vgain;
+
+  ASSERT_ALIGNED16(aInput);
+  ASSERT_ALIGNED16(aOutput);
+  ASSERT_MULTIPLE16(aSize);
+
+  vgain = _mm_load1_ps(&aScale);
+
+  for (unsigned i = 0; i < aSize; i+=16) {
+    vin0 = _mm_load_ps(&aInput[i]);
+    vin1 = _mm_load_ps(&aInput[i + 4]);
+    vin2 = _mm_load_ps(&aInput[i + 8]);
+    vin3 = _mm_load_ps(&aInput[i + 12]);
+
+    vscaled0 = _mm_mul_ps(vin0, vgain);
+    vscaled1 = _mm_mul_ps(vin1, vgain);
+    vscaled2 = _mm_mul_ps(vin2, vgain);
+    vscaled3 = _mm_mul_ps(vin3, vgain);
+
+    vin0 = _mm_load_ps(&aOutput[i]);
+    vin1 = _mm_load_ps(&aOutput[i + 4]);
+    vin2 = _mm_load_ps(&aOutput[i + 8]);
+    vin3 = _mm_load_ps(&aOutput[i + 12]);
+
+    vout0 = _mm_add_ps(vin0, vscaled0);
+    vout1 = _mm_add_ps(vin1, vscaled1);
+    vout2 = _mm_add_ps(vin2, vscaled2);
+    vout3 = _mm_add_ps(vin3, vscaled3);
+
+    _mm_store_ps(&aOutput[i], vout0);
+    _mm_store_ps(&aOutput[i + 4], vout1);
+    _mm_store_ps(&aOutput[i + 8], vout2);
+    _mm_store_ps(&aOutput[i + 12], vout3);
+  }
+}
+
+void
+AudioBlockCopyChannelWithScale_SSE(const float* aInput,
+                                   float aScale,
+                                   float* aOutput)
+{
+  __m128 vin0, vin1, vin2, vin3,
+         vout0, vout1, vout2, vout3;
+
+  ASSERT_ALIGNED16(aInput);
+  ASSERT_ALIGNED16(aOutput);
+
+  __m128 vgain = _mm_load1_ps(&aScale);
+
+  for (unsigned i = 0 ; i < WEBAUDIO_BLOCK_SIZE; i+=16) {
+    vin0 = _mm_load_ps(&aInput[i]);
+    vin1 = _mm_load_ps(&aInput[i + 4]);
+    vin2 = _mm_load_ps(&aInput[i + 8]);
+    vin3 = _mm_load_ps(&aInput[i + 12]);
+    vout0 = _mm_mul_ps(vin0, vgain);
+    vout1 = _mm_mul_ps(vin1, vgain);
+    vout2 = _mm_mul_ps(vin2, vgain);
+    vout3 = _mm_mul_ps(vin3, vgain);
+    _mm_store_ps(&aOutput[i], vout0);
+    _mm_store_ps(&aOutput[i + 4], vout1);
+    _mm_store_ps(&aOutput[i + 8], vout2);
+    _mm_store_ps(&aOutput[i + 12], vout3);
+  }
+}
+
+void
+AudioBlockCopyChannelWithScale_SSE(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                   const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                   float aOutput[WEBAUDIO_BLOCK_SIZE])
+{
+  __m128 vin0, vin1, vin2, vin3,
+         vscaled0, vscaled1, vscaled2, vscaled3,
+         vout0, vout1, vout2, vout3;
+
+  ASSERT_ALIGNED16(aInput);
+  ASSERT_ALIGNED16(aScale);
+  ASSERT_ALIGNED16(aOutput);
+
+  for (unsigned i = 0 ; i < WEBAUDIO_BLOCK_SIZE; i+=16) {
+    vscaled0 = _mm_load_ps(&aScale[i]);
+    vscaled1 = _mm_load_ps(&aScale[i+4]);
+    vscaled2 = _mm_load_ps(&aScale[i+8]);
+    vscaled3 = _mm_load_ps(&aScale[i+12]);
+
+    vin0 = _mm_load_ps(&aInput[i]);
+    vin1 = _mm_load_ps(&aInput[i + 4]);
+    vin2 = _mm_load_ps(&aInput[i + 8]);
+    vin3 = _mm_load_ps(&aInput[i + 12]);
+
+    vout0 = _mm_mul_ps(vin0, vscaled0);
+    vout1 = _mm_mul_ps(vin1, vscaled1);
+    vout2 = _mm_mul_ps(vin2, vscaled2);
+    vout3 = _mm_mul_ps(vin3, vscaled3);
+
+    _mm_store_ps(&aOutput[i], vout0);
+    _mm_store_ps(&aOutput[i + 4], vout1);
+    _mm_store_ps(&aOutput[i + 8], vout2);
+    _mm_store_ps(&aOutput[i + 12], vout3);
+  }
+}
+
+void
+AudioBufferInPlaceScale_SSE(float* aBlock,
+                            float aScale,
+                            uint32_t aSize)
+{
+  __m128 vout0, vout1, vout2, vout3,
+         vin0, vin1, vin2, vin3;
+
+  ASSERT_ALIGNED16(aBlock);
+  ASSERT_MULTIPLE16(aSize);
+
+  __m128 vgain = _mm_load1_ps(&aScale);
+
+  for (unsigned i = 0; i < aSize; i+=16) {
+    vin0 = _mm_load_ps(&aBlock[i]);
+    vin1 = _mm_load_ps(&aBlock[i + 4]);
+    vin2 = _mm_load_ps(&aBlock[i + 8]);
+    vin3 = _mm_load_ps(&aBlock[i + 12]);
+    vout0 = _mm_mul_ps(vin0, vgain);
+    vout1 = _mm_mul_ps(vin1, vgain);
+    vout2 = _mm_mul_ps(vin2, vgain);
+    vout3 = _mm_mul_ps(vin3, vgain);
+    _mm_store_ps(&aBlock[i], vout0);
+    _mm_store_ps(&aBlock[i + 4], vout1);
+    _mm_store_ps(&aBlock[i + 8], vout2);
+    _mm_store_ps(&aBlock[i + 12], vout3);
+  }
+}
+
+void
+AudioBlockPanStereoToStereo_SSE(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                float aGainL, float aGainR, bool aIsOnTheLeft,
+                                float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                float aOutputR[WEBAUDIO_BLOCK_SIZE])
+{
+  __m128 vinl0, vinr0, vinl1, vinr1,
+         vout0, vout1,
+         vscaled0, vscaled1,
+         vgainl, vgainr;
+
+  ASSERT_ALIGNED16(aInputL);
+  ASSERT_ALIGNED16(aInputR);
+  ASSERT_ALIGNED16(aOutputL);
+  ASSERT_ALIGNED16(aOutputR);
+
+  vgainl = _mm_load1_ps(&aGainL);
+  vgainr = _mm_load1_ps(&aGainR);
+
+  if (aIsOnTheLeft) {
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=8) {
+      vinl0 = _mm_load_ps(&aInputL[i]);
+      vinr0 = _mm_load_ps(&aInputR[i]);
+      vinl1 = _mm_load_ps(&aInputL[i+4]);
+      vinr1 = _mm_load_ps(&aInputR[i+4]);
+
+      /* left channel : aOutputL  = aInputL + aInputR * gainL */
+      vscaled0 = _mm_mul_ps(vinr0, vgainl);
+      vscaled1 = _mm_mul_ps(vinr1, vgainl);
+      vout0 = _mm_add_ps(vscaled0, vinl0);
+      vout1 = _mm_add_ps(vscaled1, vinl1);
+      _mm_store_ps(&aOutputL[i], vout0);
+      _mm_store_ps(&aOutputL[i+4], vout1);
+
+      /* right channel : aOutputR = aInputR * gainR */
+      vscaled0 = _mm_mul_ps(vinr0, vgainr);
+      vscaled1 = _mm_mul_ps(vinr1, vgainr);
+      _mm_store_ps(&aOutputR[i], vscaled0);
+      _mm_store_ps(&aOutputR[i+4], vscaled1);
+    }
+  } else {
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; i+=8) {
+      vinl0 = _mm_load_ps(&aInputL[i]);
+      vinr0 = _mm_load_ps(&aInputR[i]);
+      vinl1 = _mm_load_ps(&aInputL[i+4]);
+      vinr1 = _mm_load_ps(&aInputR[i+4]);
+
+      /* left channel : aInputL * gainL */
+      vscaled0 = _mm_mul_ps(vinl0, vgainl);
+      vscaled1 = _mm_mul_ps(vinl1, vgainl);
+      _mm_store_ps(&aOutputL[i], vscaled0);
+      _mm_store_ps(&aOutputL[i+4], vscaled1);
+
+      /* right channel: aOutputR = aInputR + aInputL * gainR */
+      vscaled0 = _mm_mul_ps(vinl0, vgainr);
+      vscaled1 = _mm_mul_ps(vinl1, vgainr);
+      vout0 = _mm_add_ps(vscaled0, vinr0);
+      vout1 = _mm_add_ps(vscaled1, vinr1);
+      _mm_store_ps(&aOutputR[i], vout0);
+      _mm_store_ps(&aOutputR[i+4], vout1);
+    }
+  }
+}
+
+void BufferComplexMultiply_SSE(const float* aInput,
+                               const float* aScale,
+                               float* aOutput,
+                               uint32_t aSize)
+{
+  unsigned i;
+  __m128 in0, in1, in2, in3,
+         outreal0, outreal1, outreal2, outreal3,
+         outimag0, outimag1, outimag2, outimag3;
+
+  ASSERT_ALIGNED16(aInput);
+  ASSERT_ALIGNED16(aScale);
+  ASSERT_ALIGNED16(aOutput);
+  ASSERT_MULTIPLE16(aSize);
+
+  for (i = 0; i < aSize * 2; i += 16) {
+    in0 = _mm_load_ps(&aInput[i]);
+    in1 = _mm_load_ps(&aInput[i + 4]);
+    in2 = _mm_load_ps(&aInput[i + 8]);
+    in3 = _mm_load_ps(&aInput[i + 12]);
+
+    outreal0 = _mm_shuffle_ps(in0, in1, _MM_SHUFFLE(2, 0, 2, 0));
+    outimag0 = _mm_shuffle_ps(in0, in1, _MM_SHUFFLE(3, 1, 3, 1));
+    outreal2 = _mm_shuffle_ps(in2, in3, _MM_SHUFFLE(2, 0, 2, 0));
+    outimag2 = _mm_shuffle_ps(in2, in3, _MM_SHUFFLE(3, 1, 3, 1));
+
+    in0 = _mm_load_ps(&aScale[i]);
+    in1 = _mm_load_ps(&aScale[i + 4]);
+    in2 = _mm_load_ps(&aScale[i + 8]);
+    in3 = _mm_load_ps(&aScale[i + 12]);
+
+    outreal1 = _mm_shuffle_ps(in0, in1, _MM_SHUFFLE(2, 0, 2, 0));
+    outimag1 = _mm_shuffle_ps(in0, in1, _MM_SHUFFLE(3, 1, 3, 1));
+    outreal3 = _mm_shuffle_ps(in2, in3, _MM_SHUFFLE(2, 0, 2, 0));
+    outimag3 = _mm_shuffle_ps(in2, in3, _MM_SHUFFLE(3, 1, 3, 1));
+
+    in0 = _mm_sub_ps(_mm_mul_ps(outreal0, outreal1),
+                     _mm_mul_ps(outimag0, outimag1));
+    in1 = _mm_add_ps(_mm_mul_ps(outreal0, outimag1),
+                     _mm_mul_ps(outimag0, outreal1));
+    in2 = _mm_sub_ps(_mm_mul_ps(outreal2, outreal3),
+                     _mm_mul_ps(outimag2, outimag3));
+    in3 = _mm_add_ps(_mm_mul_ps(outreal2, outimag3),
+                     _mm_mul_ps(outimag2, outreal3));
+
+    outreal0 = _mm_unpacklo_ps(in0, in1);
+    outreal1 = _mm_unpackhi_ps(in0, in1);
+    outreal2 = _mm_unpacklo_ps(in2, in3);
+    outreal3 = _mm_unpackhi_ps(in2, in3);
+
+    _mm_store_ps(&aOutput[i], outreal0);
+    _mm_store_ps(&aOutput[i + 4], outreal1);
+    _mm_store_ps(&aOutput[i + 8], outreal2);
+    _mm_store_ps(&aOutput[i + 12], outreal3);
+  }
+}
+
+float
+AudioBufferSumOfSquares_SSE(const float* aInput, uint32_t aLength)
+{
+  unsigned i;
+  __m128 in0, in1, in2, in3,
+         acc0, acc1, acc2, acc3;
+  float out[4];
+
+  ASSERT_ALIGNED16(aInput);
+  ASSERT_MULTIPLE16(aLength);
+
+  acc0 = _mm_setzero_ps();
+  acc1 = _mm_setzero_ps();
+  acc2 = _mm_setzero_ps();
+  acc3 = _mm_setzero_ps();
+
+  for (i = 0; i < aLength; i+=16) {
+    in0 = _mm_load_ps(&aInput[i]);
+    in1 = _mm_load_ps(&aInput[i + 4]);
+    in2 = _mm_load_ps(&aInput[i + 8]);
+    in3 = _mm_load_ps(&aInput[i + 12]);
+
+    in0 = _mm_mul_ps(in0, in0);
+    in1 = _mm_mul_ps(in1, in1);
+    in2 = _mm_mul_ps(in2, in2);
+    in3 = _mm_mul_ps(in3, in3);
+
+    acc0 = _mm_add_ps(acc0, in0);
+    acc1 = _mm_add_ps(acc1, in1);
+    acc2 = _mm_add_ps(acc2, in2);
+    acc3 = _mm_add_ps(acc3, in3);
+  }
+
+  acc0 = _mm_add_ps(acc0, acc1);
+  acc0 = _mm_add_ps(acc0, acc2);
+  acc0 = _mm_add_ps(acc0, acc3);
+
+  _mm_store_ps(out, acc0);
+
+  return out[0] + out[1] + out[2] + out[3];
+}
+
+}
diff --git a/dom/media/webaudio/AudioNodeEngineSSE2.h b/dom/media/webaudio/AudioNodeEngineSSE2.h
new file mode 100644
index 0000000000..d24641249a
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineSSE2.h
@@ -0,0 +1,45 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+void
+AudioBufferAddWithScale_SSE(const float* aInput,
+                            float aScale,
+                            float* aOutput,
+                            uint32_t aSize);
+
+void
+AudioBlockCopyChannelWithScale_SSE(const float* aInput,
+                                   float aScale,
+                                   float* aOutput);
+
+void
+AudioBlockCopyChannelWithScale_SSE(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                   const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                   float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+void
+AudioBufferInPlaceScale_SSE(float* aBlock,
+                            float aScale,
+                            uint32_t aSize);
+
+void
+AudioBlockPanStereoToStereo_SSE(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                float aGainL, float aGainR, bool aIsOnTheLeft,
+                                float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+float
+AudioBufferSumOfSquares_SSE(const float* aInput, uint32_t aLength);
+
+void
+BufferComplexMultiply_SSE(const float* aInput,
+                          const float* aScale,
+                          float* aOutput,
+                          uint32_t aSize);
+}
diff --git a/dom/media/webaudio/AudioNodeExternalInputStream.cpp b/dom/media/webaudio/AudioNodeExternalInputStream.cpp
new file mode 100644
index 0000000000..2dff1488b9
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeExternalInputStream.cpp
@@ -0,0 +1,238 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AlignedTArray.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputStream.h"
+#include "AudioChannelFormat.h"
+#include "mozilla/dom/MediaStreamAudioSourceNode.h"
+
+using namespace mozilla::dom;
+
+namespace mozilla {
+
+AudioNodeExternalInputStream::AudioNodeExternalInputStream(AudioNodeEngine* aEngine, TrackRate aSampleRate)
+  : AudioNodeStream(aEngine, NO_STREAM_FLAGS, aSampleRate)
+{
+  MOZ_COUNT_CTOR(AudioNodeExternalInputStream);
+}
+
+AudioNodeExternalInputStream::~AudioNodeExternalInputStream()
+{
+  MOZ_COUNT_DTOR(AudioNodeExternalInputStream);
+}
+
+/* static */ already_AddRefed<AudioNodeExternalInputStream>
+AudioNodeExternalInputStream::Create(MediaStreamGraph* aGraph,
+                                     AudioNodeEngine* aEngine)
+{
+  AudioContext* ctx = aEngine->NodeMainThread()->Context();
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aGraph->GraphRate() == ctx->SampleRate());
+
+  RefPtr<AudioNodeExternalInputStream> stream =
+    new AudioNodeExternalInputStream(aEngine, aGraph->GraphRate());
+  stream->mSuspendedCount += ctx->ShouldSuspendNewStream();
+  aGraph->AddStream(stream);
+  return stream.forget();
+}
+
+/**
+ * Copies the data in aInput to aOffsetInBlock within aBlock.
+ * aBlock must have been allocated with AllocateInputBlock and have a channel
+ * count that's a superset of the channels in aInput.
+ */
+template <typename T>
+static void
+CopyChunkToBlock(AudioChunk& aInput, AudioBlock *aBlock,
+                 uint32_t aOffsetInBlock)
+{
+  uint32_t blockChannels = aBlock->ChannelCount();
+  AutoTArray<const T*,2> channels;
+  if (aInput.IsNull()) {
+    channels.SetLength(blockChannels);
+    PodZero(channels.Elements(), blockChannels);
+  } else {
+    const nsTArray<const T*>& inputChannels = aInput.ChannelData<T>();
+    channels.SetLength(inputChannels.Length());
+    PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
+    if (channels.Length() != blockChannels) {
+      // We only need to upmix here because aBlock's channel count has been
+      // chosen to be a superset of the channel count of every chunk.
+      AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
+    }
+  }
+
+  for (uint32_t c = 0; c < blockChannels; ++c) {
+    float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
+    if (channels[c]) {
+      ConvertAudioSamplesWithScale(channels[c], outputData, aInput.GetDuration(), aInput.mVolume);
+    } else {
+      PodZero(outputData, aInput.GetDuration());
+    }
+  }
+}
+
+/**
+ * Converts the data in aSegment to a single chunk aBlock. aSegment must have
+ * duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
+ * channels in every chunk of aSegment. aBlock must be float format or null.
+ */
+static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
+                                       AudioBlock* aBlock,
+                                       int32_t aFallbackChannelCount)
+{
+  NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE, "Bad segment duration");
+
+  {
+    AudioSegment::ChunkIterator ci(*aSegment);
+    NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
+    if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
+        (ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
+
+      bool aligned = true;
+      for (size_t i = 0; i < ci->mChannelData.Length(); ++i) {
+        if (!IS_ALIGNED16(ci->mChannelData[i])) {
+            aligned = false;
+            break;
+        }
+      }
+
+      // Return this chunk directly to avoid copying data.
+      if (aligned) {
+        *aBlock = *ci;
+        return;
+      }
+    }
+  }
+
+  aBlock->AllocateChannels(aFallbackChannelCount);
+
+  uint32_t duration = 0;
+  for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
+    switch (ci->mBufferFormat) {
+      case AUDIO_FORMAT_S16: {
+        CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_FLOAT32: {
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_SILENCE: {
+        // The actual type of the sample does not matter here, but we still need
+        // to send some audio to the graph.
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+    }
+    duration += ci->GetDuration();
+  }
+}
+
+void
+AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                           uint32_t aFlags)
+{
+  // According to spec, number of outputs is always 1.
+  MOZ_ASSERT(mLastChunks.Length() == 1);
+
+  // GC stuff can result in our input stream being destroyed before this stream.
+  // Handle that.
+  if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  MOZ_ASSERT(mInputs.Length() == 1);
+
+  MediaStream* source = mInputs[0]->GetSource();
+  AutoTArray<AudioSegment,1> audioSegments;
+  uint32_t inputChannels = 0;
+  for (StreamTracks::TrackIter tracks(source->mTracks);
+       !tracks.IsEnded(); tracks.Next()) {
+    const StreamTracks::Track& inputTrack = *tracks;
+    if (!mInputs[0]->PassTrackThrough(tracks->GetID())) {
+      continue;
+    }
+
+    if (inputTrack.GetSegment()->GetType() == MediaSegment::VIDEO) {
+      MOZ_ASSERT(false, "AudioNodeExternalInputStream shouldn't have video tracks");
+      continue;
+    }
+
+    const AudioSegment& inputSegment =
+        *static_cast<AudioSegment*>(inputTrack.GetSegment());
+    if (inputSegment.IsNull()) {
+      continue;
+    }
+
+    AudioSegment& segment = *audioSegments.AppendElement();
+    GraphTime next;
+    for (GraphTime t = aFrom; t < aTo; t = next) {
+      MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
+      interval.mEnd = std::min(interval.mEnd, aTo);
+      if (interval.mStart >= interval.mEnd)
+        break;
+      next = interval.mEnd;
+
+      // We know this stream does not block during the processing interval ---
+      // we're not finished, we don't underrun, and we're not suspended.
+      StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
+      StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
+      StreamTime ticks = outputEnd - outputStart;
+
+      if (interval.mInputIsBlocked) {
+        segment.AppendNullData(ticks);
+      } else {
+        // The input stream is not blocked in this interval, so no need to call
+        // GraphTimeToStreamTimeWithBlocking.
+        StreamTime inputStart =
+          std::min(inputSegment.GetDuration(),
+                   source->GraphTimeToStreamTime(interval.mStart));
+        StreamTime inputEnd =
+          std::min(inputSegment.GetDuration(),
+                   source->GraphTimeToStreamTime(interval.mEnd));
+
+        segment.AppendSlice(inputSegment, inputStart, inputEnd);
+        // Pad if we're looking past the end of the track
+        segment.AppendNullData(ticks - (inputEnd - inputStart));
+      }
+    }
+
+    for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded(); iter.Next()) {
+      inputChannels = GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
+    }
+  }
+
+  uint32_t accumulateIndex = 0;
+  if (inputChannels) {
+    DownmixBufferType downmixBuffer;
+    ASSERT_ALIGNED16(downmixBuffer.Elements());
+    for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
+      AudioBlock tmpChunk;
+      ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk, inputChannels);
+      if (!tmpChunk.IsNull()) {
+        if (accumulateIndex == 0) {
+          mLastChunks[0].AllocateChannels(inputChannels);
+        }
+        AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
+        accumulateIndex++;
+      }
+    }
+  }
+  if (accumulateIndex == 0) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+bool
+AudioNodeExternalInputStream::IsEnabled()
+{
+  return ((MediaStreamAudioSourceNodeEngine*)Engine())->IsEnabled();
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeExternalInputStream.h b/dom/media/webaudio/AudioNodeExternalInputStream.h
new file mode 100644
index 0000000000..83d2bba749
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeExternalInputStream.h
@@ -0,0 +1,45 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODEEXTERNALINPUTSTREAM_H_
+#define MOZILLA_AUDIONODEEXTERNALINPUTSTREAM_H_
+
+#include "MediaStreamGraph.h"
+#include "AudioNodeStream.h"
+#include "mozilla/Atomics.h"
+
+namespace mozilla {
+
+/**
+ * This is a MediaStream implementation that acts for a Web Audio node but
+ * unlike other AudioNodeStreams, supports any kind of MediaStream as an
+ * input --- handling any number of audio tracks and handling blocking of
+ * the input MediaStream.
+ */
+class AudioNodeExternalInputStream final : public AudioNodeStream
+{
+public:
+  static already_AddRefed<AudioNodeExternalInputStream>
+  Create(MediaStreamGraph* aGraph, AudioNodeEngine* aEngine);
+
+protected:
+  AudioNodeExternalInputStream(AudioNodeEngine* aEngine, TrackRate aSampleRate);
+  ~AudioNodeExternalInputStream();
+
+public:
+  void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+
+private:
+  /**
+   * Determines if this is enabled or not.  Disabled nodes produce silence.
+   * This node becomes disabled if the document principal does not subsume the
+   * DOMMediaStream principal.
+   */
+  bool IsEnabled();
+};
+
+} // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODESTREAM_H_ */
diff --git a/dom/media/webaudio/AudioNodeStream.cpp b/dom/media/webaudio/AudioNodeStream.cpp
new file mode 100644
index 0000000000..0e5aa3fc73
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeStream.cpp
@@ -0,0 +1,783 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeStream.h"
+
+#include "MediaStreamGraphImpl.h"
+#include "MediaStreamListener.h"
+#include "AudioNodeEngine.h"
+#include "ThreeDPoint.h"
+#include "AudioChannelFormat.h"
+#include "AudioParamTimeline.h"
+#include "AudioContext.h"
+#include "nsMathUtils.h"
+
+using namespace mozilla::dom;
+
+namespace mozilla {
+
+/**
+ * An AudioNodeStream produces a single audio track with ID
+ * AUDIO_TRACK. This track has rate AudioContext::sIdealAudioRate
+ * for regular audio contexts, and the rate requested by the web content
+ * for offline audio contexts.
+ * Each chunk in the track is a single block of WEBAUDIO_BLOCK_SIZE samples.
+ * Note: This must be a different value than MEDIA_STREAM_DEST_TRACK_ID
+ */
+
+AudioNodeStream::AudioNodeStream(AudioNodeEngine* aEngine,
+                                 Flags aFlags,
+                                 TrackRate aSampleRate)
+  : ProcessedMediaStream(),
+    mEngine(aEngine),
+    mSampleRate(aSampleRate),
+    mFlags(aFlags),
+    mNumberOfInputChannels(2),
+    mIsActive(aEngine->IsActive()),
+    mMarkAsFinishedAfterThisBlock(false),
+    mAudioParamStream(false),
+    mPassThrough(false)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  mSuspendedCount = !(mIsActive || mFlags & EXTERNAL_OUTPUT);
+  mChannelCountMode = ChannelCountMode::Max;
+  mChannelInterpretation = ChannelInterpretation::Speakers;
+  // AudioNodes are always producing data
+  mHasCurrentData = true;
+  mLastChunks.SetLength(std::max(uint16_t(1), mEngine->OutputCount()));
+  MOZ_COUNT_CTOR(AudioNodeStream);
+}
+
+AudioNodeStream::~AudioNodeStream()
+{
+  MOZ_ASSERT(mActiveInputCount == 0);
+  MOZ_COUNT_DTOR(AudioNodeStream);
+}
+
+void
+AudioNodeStream::DestroyImpl()
+{
+  // These are graph thread objects, so clean up on graph thread.
+  mInputChunks.Clear();
+  mLastChunks.Clear();
+
+  ProcessedMediaStream::DestroyImpl();
+}
+
+/* static */ already_AddRefed<AudioNodeStream>
+AudioNodeStream::Create(AudioContext* aCtx, AudioNodeEngine* aEngine,
+                        Flags aFlags, MediaStreamGraph* aGraph)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_RELEASE_ASSERT(aGraph);
+
+  // MediaRecorders use an AudioNodeStream, but no AudioNode
+  AudioNode* node = aEngine->NodeMainThread();
+
+  RefPtr<AudioNodeStream> stream =
+    new AudioNodeStream(aEngine, aFlags, aGraph->GraphRate());
+  stream->mSuspendedCount += aCtx->ShouldSuspendNewStream();
+  if (node) {
+    stream->SetChannelMixingParametersImpl(node->ChannelCount(),
+                                           node->ChannelCountModeValue(),
+                                           node->ChannelInterpretationValue());
+  }
+  aGraph->AddStream(stream);
+  return stream.forget();
+}
+
+size_t
+AudioNodeStream::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = 0;
+
+  // Not reported:
+  // - mEngine
+
+  amount += ProcessedMediaStream::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mLastChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mLastChunks.Length(); i++) {
+    // NB: This is currently unshared only as there are instances of
+    //     double reporting in DMD otherwise.
+    amount += mLastChunks[i].SizeOfExcludingThisIfUnshared(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t
+AudioNodeStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void
+AudioNodeStream::SizeOfAudioNodesIncludingThis(MallocSizeOf aMallocSizeOf,
+                                               AudioNodeSizes& aUsage) const
+{
+  // Explicitly separate out the stream memory.
+  aUsage.mStream = SizeOfIncludingThis(aMallocSizeOf);
+
+  if (mEngine) {
+    // This will fill out the rest of |aUsage|.
+    mEngine->SizeOfIncludingThis(aMallocSizeOf, aUsage);
+  }
+}
+
+void
+AudioNodeStream::SetStreamTimeParameter(uint32_t aIndex, AudioContext* aContext,
+                                        double aStreamTime)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, uint32_t aIndex, MediaStream* aRelativeToStream,
+            double aStreamTime)
+      : ControlMessage(aStream), mStreamTime(aStreamTime),
+        mRelativeToStream(aRelativeToStream), mIndex(aIndex)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->
+          SetStreamTimeParameterImpl(mIndex, mRelativeToStream, mStreamTime);
+    }
+    double mStreamTime;
+    MediaStream* mRelativeToStream;
+    uint32_t mIndex;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aIndex,
+                                                 aContext->DestinationStream(),
+                                                 aStreamTime));
+}
+
+void
+AudioNodeStream::SetStreamTimeParameterImpl(uint32_t aIndex, MediaStream* aRelativeToStream,
+                                            double aStreamTime)
+{
+  StreamTime ticks = aRelativeToStream->SecondsToNearestStreamTime(aStreamTime);
+  mEngine->SetStreamTimeParameter(aIndex, ticks);
+}
+
+void
+AudioNodeStream::SetDoubleParameter(uint32_t aIndex, double aValue)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, uint32_t aIndex, double aValue)
+      : ControlMessage(aStream), mValue(aValue), mIndex(aIndex)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->
+          SetDoubleParameter(mIndex, mValue);
+    }
+    double mValue;
+    uint32_t mIndex;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aIndex, aValue));
+}
+
+void
+AudioNodeStream::SetInt32Parameter(uint32_t aIndex, int32_t aValue)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, uint32_t aIndex, int32_t aValue)
+      : ControlMessage(aStream), mValue(aValue), mIndex(aIndex)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->
+          SetInt32Parameter(mIndex, mValue);
+    }
+    int32_t mValue;
+    uint32_t mIndex;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aIndex, aValue));
+}
+
+void
+AudioNodeStream::SendTimelineEvent(uint32_t aIndex,
+                                   const AudioTimelineEvent& aEvent)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, uint32_t aIndex,
+            const AudioTimelineEvent& aEvent)
+      : ControlMessage(aStream),
+        mEvent(aEvent),
+        mSampleRate(aStream->SampleRate()),
+        mIndex(aIndex)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->
+          RecvTimelineEvent(mIndex, mEvent);
+    }
+    AudioTimelineEvent mEvent;
+    TrackRate mSampleRate;
+    uint32_t mIndex;
+  };
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aIndex, aEvent));
+}
+
+void
+AudioNodeStream::SetThreeDPointParameter(uint32_t aIndex, const ThreeDPoint& aValue)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, uint32_t aIndex, const ThreeDPoint& aValue)
+      : ControlMessage(aStream), mValue(aValue), mIndex(aIndex)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->
+          SetThreeDPointParameter(mIndex, mValue);
+    }
+    ThreeDPoint mValue;
+    uint32_t mIndex;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aIndex, aValue));
+}
+
+void
+AudioNodeStream::SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList>&& aBuffer)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream,
+            already_AddRefed<ThreadSharedFloatArrayBufferList>& aBuffer)
+      : ControlMessage(aStream), mBuffer(aBuffer)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->
+          SetBuffer(mBuffer.forget());
+    }
+    RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aBuffer));
+}
+
+void
+AudioNodeStream::SetRawArrayData(nsTArray<float>& aData)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream,
+            nsTArray<float>& aData)
+      : ControlMessage(aStream)
+    {
+      mData.SwapElements(aData);
+    }
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->Engine()->SetRawArrayData(mData);
+    }
+    nsTArray<float> mData;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aData));
+}
+
+void
+AudioNodeStream::SetChannelMixingParameters(uint32_t aNumberOfChannels,
+                                            ChannelCountMode aChannelCountMode,
+                                            ChannelInterpretation aChannelInterpretation)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream,
+            uint32_t aNumberOfChannels,
+            ChannelCountMode aChannelCountMode,
+            ChannelInterpretation aChannelInterpretation)
+      : ControlMessage(aStream),
+        mNumberOfChannels(aNumberOfChannels),
+        mChannelCountMode(aChannelCountMode),
+        mChannelInterpretation(aChannelInterpretation)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->
+        SetChannelMixingParametersImpl(mNumberOfChannels, mChannelCountMode,
+                                       mChannelInterpretation);
+    }
+    uint32_t mNumberOfChannels;
+    ChannelCountMode mChannelCountMode;
+    ChannelInterpretation mChannelInterpretation;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aNumberOfChannels,
+                                                 aChannelCountMode,
+                                                 aChannelInterpretation));
+}
+
+void
+AudioNodeStream::SetPassThrough(bool aPassThrough)
+{
+  class Message final : public ControlMessage
+  {
+  public:
+    Message(AudioNodeStream* aStream, bool aPassThrough)
+      : ControlMessage(aStream), mPassThrough(aPassThrough)
+    {}
+    void Run() override
+    {
+      static_cast<AudioNodeStream*>(mStream)->mPassThrough = mPassThrough;
+    }
+    bool mPassThrough;
+  };
+
+  GraphImpl()->AppendMessage(MakeUnique<Message>(this, aPassThrough));
+}
+
+void
+AudioNodeStream::SetChannelMixingParametersImpl(uint32_t aNumberOfChannels,
+                                                ChannelCountMode aChannelCountMode,
+                                                ChannelInterpretation aChannelInterpretation)
+{
+  // Make sure that we're not clobbering any significant bits by fitting these
+  // values in 16 bits.
+  MOZ_ASSERT(int(aChannelCountMode) < INT16_MAX);
+  MOZ_ASSERT(int(aChannelInterpretation) < INT16_MAX);
+
+  mNumberOfInputChannels = aNumberOfChannels;
+  mChannelCountMode = aChannelCountMode;
+  mChannelInterpretation = aChannelInterpretation;
+}
+
+uint32_t
+AudioNodeStream::ComputedNumberOfChannels(uint32_t aInputChannelCount)
+{
+  switch (mChannelCountMode) {
+  case ChannelCountMode::Explicit:
+    // Disregard the channel count we've calculated from inputs, and just use
+    // mNumberOfInputChannels.
+    return mNumberOfInputChannels;
+  case ChannelCountMode::Clamped_max:
+    // Clamp the computed output channel count to mNumberOfInputChannels.
+    return std::min(aInputChannelCount, mNumberOfInputChannels);
+  default:
+  case ChannelCountMode::Max:
+    // Nothing to do here, just shut up the compiler warning.
+    return aInputChannelCount;
+  }
+}
+
+class AudioNodeStream::AdvanceAndResumeMessage final : public ControlMessage {
+public:
+  AdvanceAndResumeMessage(AudioNodeStream* aStream, StreamTime aAdvance) :
+    ControlMessage(aStream), mAdvance(aAdvance) {}
+  void Run() override
+  {
+    auto ns = static_cast<AudioNodeStream*>(mStream);
+    ns->mTracksStartTime -= mAdvance;
+
+    StreamTracks::Track* track = ns->EnsureTrack(AUDIO_TRACK);
+    track->Get<AudioSegment>()->AppendNullData(mAdvance);
+
+    ns->GraphImpl()->DecrementSuspendCount(mStream);
+  }
+private:
+  StreamTime mAdvance;
+};
+
+void
+AudioNodeStream::AdvanceAndResume(StreamTime aAdvance)
+{
+  mMainThreadCurrentTime += aAdvance;
+  GraphImpl()->AppendMessage(MakeUnique<AdvanceAndResumeMessage>(this, aAdvance));
+}
+
+void
+AudioNodeStream::ObtainInputBlock(AudioBlock& aTmpChunk,
+                                  uint32_t aPortIndex)
+{
+  uint32_t inputCount = mInputs.Length();
+  uint32_t outputChannelCount = 1;
+  AutoTArray<const AudioBlock*,250> inputChunks;
+  for (uint32_t i = 0; i < inputCount; ++i) {
+    if (aPortIndex != mInputs[i]->InputNumber()) {
+      // This input is connected to a different port
+      continue;
+    }
+    MediaStream* s = mInputs[i]->GetSource();
+    AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
+    MOZ_ASSERT(a == s->AsAudioNodeStream());
+    if (a->IsAudioParamStream()) {
+      continue;
+    }
+
+    const AudioBlock* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
+    MOZ_ASSERT(chunk);
+    if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
+      continue;
+    }
+
+    inputChunks.AppendElement(chunk);
+    outputChannelCount =
+      GetAudioChannelsSuperset(outputChannelCount, chunk->ChannelCount());
+  }
+
+  outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
+
+  uint32_t inputChunkCount = inputChunks.Length();
+  if (inputChunkCount == 0 ||
+      (inputChunkCount == 1 && inputChunks[0]->ChannelCount() == 0)) {
+    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  if (inputChunkCount == 1 &&
+      inputChunks[0]->ChannelCount() == outputChannelCount) {
+    aTmpChunk = *inputChunks[0];
+    return;
+  }
+
+  if (outputChannelCount == 0) {
+    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  aTmpChunk.AllocateChannels(outputChannelCount);
+  DownmixBufferType downmixBuffer;
+  ASSERT_ALIGNED16(downmixBuffer.Elements());
+
+  for (uint32_t i = 0; i < inputChunkCount; ++i) {
+    AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
+  }
+}
+
+void
+AudioNodeStream::AccumulateInputChunk(uint32_t aInputIndex,
+                                      const AudioBlock& aChunk,
+                                      AudioBlock* aBlock,
+                                      DownmixBufferType* aDownmixBuffer)
+{
+  AutoTArray<const float*,GUESS_AUDIO_CHANNELS> channels;
+  UpMixDownMixChunk(&aChunk, aBlock->ChannelCount(), channels, *aDownmixBuffer);
+
+  for (uint32_t c = 0; c < channels.Length(); ++c) {
+    const float* inputData = static_cast<const float*>(channels[c]);
+    float* outputData = aBlock->ChannelFloatsForWrite(c);
+    if (inputData) {
+      if (aInputIndex == 0) {
+        AudioBlockCopyChannelWithScale(inputData, aChunk.mVolume, outputData);
+      } else {
+        AudioBlockAddChannelWithScale(inputData, aChunk.mVolume, outputData);
+      }
+    } else {
+      if (aInputIndex == 0) {
+        PodZero(outputData, WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+}
+
+void
+AudioNodeStream::UpMixDownMixChunk(const AudioBlock* aChunk,
+                                   uint32_t aOutputChannelCount,
+                                   nsTArray<const float*>& aOutputChannels,
+                                   DownmixBufferType& aDownmixBuffer)
+{
+  for (uint32_t i = 0; i < aChunk->ChannelCount(); i++) {
+    aOutputChannels.AppendElement(static_cast<const float*>(aChunk->mChannelData[i]));
+  }
+  if (aOutputChannels.Length() < aOutputChannelCount) {
+    if (mChannelInterpretation == ChannelInterpretation::Speakers) {
+      AudioChannelsUpMix<float>(&aOutputChannels, aOutputChannelCount, nullptr);
+      NS_ASSERTION(aOutputChannelCount == aOutputChannels.Length(),
+                   "We called GetAudioChannelsSuperset to avoid this");
+    } else {
+      // Fill up the remaining aOutputChannels by zeros
+      for (uint32_t j = aOutputChannels.Length(); j < aOutputChannelCount; ++j) {
+        aOutputChannels.AppendElement(nullptr);
+      }
+    }
+  } else if (aOutputChannels.Length() > aOutputChannelCount) {
+    if (mChannelInterpretation == ChannelInterpretation::Speakers) {
+      AutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
+      outputChannels.SetLength(aOutputChannelCount);
+      aDownmixBuffer.SetLength(aOutputChannelCount * WEBAUDIO_BLOCK_SIZE);
+      for (uint32_t j = 0; j < aOutputChannelCount; ++j) {
+        outputChannels[j] = &aDownmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
+      }
+
+      AudioChannelsDownMix(aOutputChannels, outputChannels.Elements(),
+                           aOutputChannelCount, WEBAUDIO_BLOCK_SIZE);
+
+      aOutputChannels.SetLength(aOutputChannelCount);
+      for (uint32_t j = 0; j < aOutputChannels.Length(); ++j) {
+        aOutputChannels[j] = outputChannels[j];
+      }
+    } else {
+      // Drop the remaining aOutputChannels
+      aOutputChannels.RemoveElementsAt(aOutputChannelCount,
+        aOutputChannels.Length() - aOutputChannelCount);
+    }
+  }
+}
+
+// The MediaStreamGraph guarantees that this is actually one block, for
+// AudioNodeStreams.
+void
+AudioNodeStream::ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags)
+{
+  uint16_t outputCount = mLastChunks.Length();
+  MOZ_ASSERT(outputCount == std::max(uint16_t(1), mEngine->OutputCount()));
+
+  if (!mIsActive) {
+    // mLastChunks are already null.
+#ifdef DEBUG
+    for (const auto& chunk : mLastChunks) {
+      MOZ_ASSERT(chunk.IsNull());
+    }
+#endif
+  } else if (InMutedCycle()) {
+    mInputChunks.Clear();
+    for (uint16_t i = 0; i < outputCount; ++i) {
+      mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+    }
+  } else {
+    // We need to generate at least one input
+    uint16_t maxInputs = std::max(uint16_t(1), mEngine->InputCount());
+    mInputChunks.SetLength(maxInputs);
+    for (uint16_t i = 0; i < maxInputs; ++i) {
+      ObtainInputBlock(mInputChunks[i], i);
+    }
+    bool finished = false;
+    if (mPassThrough) {
+      MOZ_ASSERT(outputCount == 1, "For now, we only support nodes that have one output port");
+      mLastChunks[0] = mInputChunks[0];
+    } else {
+      if (maxInputs <= 1 && outputCount <= 1) {
+        mEngine->ProcessBlock(this, aFrom,
+                              mInputChunks[0], &mLastChunks[0], &finished);
+      } else {
+        mEngine->ProcessBlocksOnPorts(this, mInputChunks, mLastChunks, &finished);
+      }
+    }
+    for (uint16_t i = 0; i < outputCount; ++i) {
+      NS_ASSERTION(mLastChunks[i].GetDuration() == WEBAUDIO_BLOCK_SIZE,
+                   "Invalid WebAudio chunk size");
+    }
+    if (finished) {
+      mMarkAsFinishedAfterThisBlock = true;
+      if (mIsActive) {
+        ScheduleCheckForInactive();
+      }
+    }
+
+    if (GetDisabledTrackMode(static_cast<TrackID>(AUDIO_TRACK)) != DisabledTrackMode::ENABLED) {
+      for (uint32_t i = 0; i < outputCount; ++i) {
+        mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+
+  if (!mFinished) {
+    // Don't output anything while finished
+    if (mFlags & EXTERNAL_OUTPUT) {
+      AdvanceOutputSegment();
+    }
+    if (mMarkAsFinishedAfterThisBlock && (aFlags & ALLOW_FINISH)) {
+      // This stream was finished the last time that we looked at it, and all
+      // of the depending streams have finished their output as well, so now
+      // it's time to mark this stream as finished.
+      if (mFlags & EXTERNAL_OUTPUT) {
+        FinishOutput();
+      }
+      FinishOnGraphThread();
+    }
+  }
+}
+
+void
+AudioNodeStream::ProduceOutputBeforeInput(GraphTime aFrom)
+{
+  MOZ_ASSERT(mEngine->AsDelayNodeEngine());
+  MOZ_ASSERT(mEngine->OutputCount() == 1,
+             "DelayNodeEngine output count should be 1");
+  MOZ_ASSERT(!InMutedCycle(), "DelayNodes should break cycles");
+  MOZ_ASSERT(mLastChunks.Length() == 1);
+
+  if (!mIsActive) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+  } else {
+    mEngine->ProduceBlockBeforeInput(this, aFrom, &mLastChunks[0]);
+    NS_ASSERTION(mLastChunks[0].GetDuration() == WEBAUDIO_BLOCK_SIZE,
+                 "Invalid WebAudio chunk size");
+    if (GetDisabledTrackMode(static_cast<TrackID>(AUDIO_TRACK)) != DisabledTrackMode::ENABLED) {
+      mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+    }
+  }
+}
+
+void
+AudioNodeStream::AdvanceOutputSegment()
+{
+  StreamTracks::Track* track = EnsureTrack(AUDIO_TRACK);
+  // No more tracks will be coming
+  mTracks.AdvanceKnownTracksTime(STREAM_TIME_MAX);
+
+  AudioSegment* segment = track->Get<AudioSegment>();
+
+  if (!mLastChunks[0].IsNull()) {
+    segment->AppendAndConsumeChunk(mLastChunks[0].AsMutableChunk());
+  } else {
+    segment->AppendNullData(mLastChunks[0].GetDuration());
+  }
+
+  for (uint32_t j = 0; j < mListeners.Length(); ++j) {
+    MediaStreamListener* l = mListeners[j];
+    AudioChunk copyChunk = mLastChunks[0].AsAudioChunk();
+    AudioSegment tmpSegment;
+    tmpSegment.AppendAndConsumeChunk(&copyChunk);
+    l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
+                                segment->GetDuration(), TrackEventCommand::TRACK_EVENT_NONE, tmpSegment);
+  }
+}
+
+void
+AudioNodeStream::FinishOutput()
+{
+  StreamTracks::Track* track = EnsureTrack(AUDIO_TRACK);
+  track->SetEnded();
+
+  for (uint32_t j = 0; j < mListeners.Length(); ++j) {
+    MediaStreamListener* l = mListeners[j];
+    AudioSegment emptySegment;
+    l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
+                                track->GetSegment()->GetDuration(),
+                                TrackEventCommand::TRACK_EVENT_ENDED, emptySegment);
+  }
+}
+
+void
+AudioNodeStream::AddInput(MediaInputPort* aPort)
+{
+  ProcessedMediaStream::AddInput(aPort);
+  AudioNodeStream* ns = aPort->GetSource()->AsAudioNodeStream();
+  // Streams that are not AudioNodeStreams are considered active.
+  if (!ns || (ns->mIsActive && !ns->IsAudioParamStream())) {
+    IncrementActiveInputCount();
+  }
+}
+void
+AudioNodeStream::RemoveInput(MediaInputPort* aPort)
+{
+  ProcessedMediaStream::RemoveInput(aPort);
+  AudioNodeStream* ns = aPort->GetSource()->AsAudioNodeStream();
+  // Streams that are not AudioNodeStreams are considered active.
+  if (!ns || (ns->mIsActive && !ns->IsAudioParamStream())) {
+    DecrementActiveInputCount();
+  }
+}
+
+void
+AudioNodeStream::SetActive()
+{
+  if (mIsActive || mMarkAsFinishedAfterThisBlock) {
+    return;
+  }
+
+  mIsActive = true;
+  if (!(mFlags & EXTERNAL_OUTPUT)) {
+    GraphImpl()->DecrementSuspendCount(this);
+  }
+  if (IsAudioParamStream()) {
+    // Consumers merely influence stream order.
+    // They do not read from the stream.
+    return;
+  }
+
+  for (const auto& consumer : mConsumers) {
+    AudioNodeStream* ns = consumer->GetDestination()->AsAudioNodeStream();
+    if (ns) {
+      ns->IncrementActiveInputCount();
+    }
+  }
+}
+
+class AudioNodeStream::CheckForInactiveMessage final : public ControlMessage
+{
+public:
+  explicit CheckForInactiveMessage(AudioNodeStream* aStream) :
+    ControlMessage(aStream) {}
+  void Run() override
+  {
+    auto ns = static_cast<AudioNodeStream*>(mStream);
+    ns->CheckForInactive();
+  }
+};
+
+void
+AudioNodeStream::ScheduleCheckForInactive()
+{
+  if (mActiveInputCount > 0 && !mMarkAsFinishedAfterThisBlock) {
+    return;
+  }
+
+  auto message = MakeUnique<CheckForInactiveMessage>(this);
+  GraphImpl()->RunMessageAfterProcessing(Move(message));
+}
+
+void
+AudioNodeStream::CheckForInactive()
+{
+  if (((mActiveInputCount > 0 || mEngine->IsActive()) &&
+       !mMarkAsFinishedAfterThisBlock) ||
+      !mIsActive) {
+    return;
+  }
+
+  mIsActive = false;
+  mInputChunks.Clear(); // not required for foreseeable future
+  for (auto& chunk : mLastChunks) {
+    chunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+  if (!(mFlags & EXTERNAL_OUTPUT)) {
+    GraphImpl()->IncrementSuspendCount(this);
+  }
+  if (IsAudioParamStream()) {
+    return;
+  }
+
+  for (const auto& consumer : mConsumers) {
+    AudioNodeStream* ns = consumer->GetDestination()->AsAudioNodeStream();
+    if (ns) {
+      ns->DecrementActiveInputCount();
+    }
+  }
+}
+
+void
+AudioNodeStream::IncrementActiveInputCount()
+{
+  ++mActiveInputCount;
+  SetActive();
+}
+
+void
+AudioNodeStream::DecrementActiveInputCount()
+{
+  MOZ_ASSERT(mActiveInputCount > 0);
+  --mActiveInputCount;
+  CheckForInactive();
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeStream.h b/dom/media/webaudio/AudioNodeStream.h
new file mode 100644
index 0000000000..87f6fa2219
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeStream.h
@@ -0,0 +1,239 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODESTREAM_H_
+#define MOZILLA_AUDIONODESTREAM_H_
+
+#include "MediaStreamGraph.h"
+#include "mozilla/dom/AudioNodeBinding.h"
+#include "nsAutoPtr.h"
+#include "AlignedTArray.h"
+#include "AudioBlock.h"
+#include "AudioSegment.h"
+
+namespace mozilla {
+
+namespace dom {
+struct ThreeDPoint;
+struct AudioTimelineEvent;
+class AudioContext;
+} // namespace dom
+
+class ThreadSharedFloatArrayBufferList;
+class AudioNodeEngine;
+
+typedef AlignedAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE, 16> DownmixBufferType;
+
+/**
+ * An AudioNodeStream produces one audio track with ID AUDIO_TRACK.
+ * The start time of the AudioTrack is aligned to the start time of the
+ * AudioContext's destination node stream, plus some multiple of BLOCK_SIZE
+ * samples.
+ *
+ * An AudioNodeStream has an AudioNodeEngine plugged into it that does the
+ * actual audio processing. AudioNodeStream contains the glue code that
+ * integrates audio processing with the MediaStreamGraph.
+ */
+class AudioNodeStream : public ProcessedMediaStream
+{
+  typedef dom::ChannelCountMode ChannelCountMode;
+  typedef dom::ChannelInterpretation ChannelInterpretation;
+
+public:
+  typedef mozilla::dom::AudioContext AudioContext;
+
+  enum { AUDIO_TRACK = 1 };
+
+  typedef AutoTArray<AudioBlock, 1> OutputChunks;
+
+  // Flags re main thread updates and stream output.
+  typedef unsigned Flags;
+  enum : Flags {
+    NO_STREAM_FLAGS = 0U,
+    NEED_MAIN_THREAD_FINISHED = 1U << 0,
+    NEED_MAIN_THREAD_CURRENT_TIME = 1U << 1,
+    // Internal AudioNodeStreams can only pass their output to another
+    // AudioNode, whereas external AudioNodeStreams can pass their output
+    // to other ProcessedMediaStreams or hardware audio output.
+    EXTERNAL_OUTPUT = 1U << 2,
+  };
+  /**
+   * Create a stream that will process audio for an AudioNode.
+   * Takes ownership of aEngine.
+   * aGraph is required and equals the graph of aCtx in most cases. An exception
+   * is AudioDestinationNode where the context's graph hasn't been set up yet.
+   */
+  static already_AddRefed<AudioNodeStream>
+  Create(AudioContext* aCtx, AudioNodeEngine* aEngine, Flags aKind,
+         MediaStreamGraph* aGraph);
+
+protected:
+  /**
+   * Transfers ownership of aEngine to the new AudioNodeStream.
+   */
+  AudioNodeStream(AudioNodeEngine* aEngine,
+                  Flags aFlags,
+                  TrackRate aSampleRate);
+
+  ~AudioNodeStream();
+
+public:
+  // Control API
+  /**
+   * Sets a parameter that's a time relative to some stream's played time.
+   * This time is converted to a time relative to this stream when it's set.
+   */
+  void SetStreamTimeParameter(uint32_t aIndex, AudioContext* aContext,
+                              double aStreamTime);
+  void SetDoubleParameter(uint32_t aIndex, double aValue);
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue);
+  void SetThreeDPointParameter(uint32_t aIndex, const dom::ThreeDPoint& aValue);
+  void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList>&& aBuffer);
+  // This sends a single event to the timeline on the MSG thread side.
+  void SendTimelineEvent(uint32_t aIndex, const dom::AudioTimelineEvent& aEvent);
+  // This consumes the contents of aData.  aData will be emptied after this returns.
+  void SetRawArrayData(nsTArray<float>& aData);
+  void SetChannelMixingParameters(uint32_t aNumberOfChannels,
+                                  ChannelCountMode aChannelCountMoe,
+                                  ChannelInterpretation aChannelInterpretation);
+  void SetPassThrough(bool aPassThrough);
+  ChannelInterpretation GetChannelInterpretation()
+  {
+    return mChannelInterpretation;
+  }
+
+  void SetAudioParamHelperStream()
+  {
+    MOZ_ASSERT(!mAudioParamStream, "Can only do this once");
+    mAudioParamStream = true;
+  }
+
+  /*
+   * Resume stream after updating its concept of current time by aAdvance.
+   * Main thread.  Used only from AudioDestinationNode when resuming a stream
+   * suspended to save running the MediaStreamGraph when there are no other
+   * nodes in the AudioContext.
+   */
+  void AdvanceAndResume(StreamTime aAdvance);
+
+  AudioNodeStream* AsAudioNodeStream() override { return this; }
+  void AddInput(MediaInputPort* aPort) override;
+  void RemoveInput(MediaInputPort* aPort) override;
+
+  // Graph thread only
+  void SetStreamTimeParameterImpl(uint32_t aIndex, MediaStream* aRelativeToStream,
+                                  double aStreamTime);
+  void SetChannelMixingParametersImpl(uint32_t aNumberOfChannels,
+                                      ChannelCountMode aChannelCountMoe,
+                                      ChannelInterpretation aChannelInterpretation);
+  void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+  /**
+   * Produce the next block of output, before input is provided.
+   * ProcessInput() will be called later, and it then should not change
+   * the output.  This is used only for DelayNodeEngine in a feedback loop.
+   */
+  void ProduceOutputBeforeInput(GraphTime aFrom);
+  bool IsAudioParamStream() const
+  {
+    return mAudioParamStream;
+  }
+
+  const OutputChunks& LastChunks() const
+  {
+    return mLastChunks;
+  }
+  bool MainThreadNeedsUpdates() const override
+  {
+    return ((mFlags & NEED_MAIN_THREAD_FINISHED) && mFinished) ||
+      (mFlags & NEED_MAIN_THREAD_CURRENT_TIME);
+  }
+
+  // Any thread
+  AudioNodeEngine* Engine() { return mEngine; }
+  TrackRate SampleRate() const { return mSampleRate; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void SizeOfAudioNodesIncludingThis(MallocSizeOf aMallocSizeOf,
+                                     AudioNodeSizes& aUsage) const;
+
+  /*
+   * SetActive() is called when either an active input is added or the engine
+   * for a source node transitions from inactive to active.  This is not
+   * called from engines for processing nodes because they only become active
+   * when there are active input streams, in which case this stream is already
+   * active.
+   */
+  void SetActive();
+  /*
+   * ScheduleCheckForInactive() is called during stream processing when the
+   * engine transitions from active to inactive, or the stream finishes.  It
+   * schedules a call to CheckForInactive() after stream processing.
+   */
+  void ScheduleCheckForInactive();
+
+protected:
+  class AdvanceAndResumeMessage;
+  class CheckForInactiveMessage;
+
+  void DestroyImpl() override;
+
+  /*
+   * CheckForInactive() is called when the engine transitions from active to
+   * inactive, or an active input is removed, or the stream finishes.  If the
+   * stream is now inactive, then mInputChunks will be cleared and mLastChunks
+   * will be set to null.  ProcessBlock() will not be called on the engine
+   * again until SetActive() is called.
+   */
+  void CheckForInactive();
+
+  void AdvanceOutputSegment();
+  void FinishOutput();
+  void AccumulateInputChunk(uint32_t aInputIndex, const AudioBlock& aChunk,
+                            AudioBlock* aBlock,
+                            DownmixBufferType* aDownmixBuffer);
+  void UpMixDownMixChunk(const AudioBlock* aChunk, uint32_t aOutputChannelCount,
+                         nsTArray<const float*>& aOutputChannels,
+                         DownmixBufferType& aDownmixBuffer);
+
+  uint32_t ComputedNumberOfChannels(uint32_t aInputChannelCount);
+  void ObtainInputBlock(AudioBlock& aTmpChunk, uint32_t aPortIndex);
+  void IncrementActiveInputCount();
+  void DecrementActiveInputCount();
+
+  // The engine that will generate output for this node.
+  nsAutoPtr<AudioNodeEngine> mEngine;
+  // The mixed input blocks are kept from iteration to iteration to avoid
+  // reallocating channel data arrays and any buffers for mixing.
+  OutputChunks mInputChunks;
+  // The last block produced by this node.
+  OutputChunks mLastChunks;
+  // The stream's sampling rate
+  const TrackRate mSampleRate;
+  // Whether this is an internal or external stream
+  const Flags mFlags;
+  // The number of input streams that may provide non-silent input.
+  uint32_t mActiveInputCount = 0;
+  // The number of input channels that this stream requires. 0 means don't care.
+  uint32_t mNumberOfInputChannels;
+  // The mixing modes
+  ChannelCountMode mChannelCountMode;
+  ChannelInterpretation mChannelInterpretation;
+  // Streams are considered active if the stream has not finished and either
+  // the engine is active or there are active input streams.
+  bool mIsActive;
+  // Whether the stream should be marked as finished as soon
+  // as the current time range has been computed block by block.
+  bool mMarkAsFinishedAfterThisBlock;
+  // Whether the stream is an AudioParamHelper stream.
+  bool mAudioParamStream;
+  // Whether the stream just passes its input through.
+  bool mPassThrough;
+};
+
+} // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODESTREAM_H_ */
diff --git a/dom/media/webaudio/AudioParam.cpp b/dom/media/webaudio/AudioParam.cpp
new file mode 100644
index 0000000000..6f55749931
--- /dev/null
+++ b/dom/media/webaudio/AudioParam.cpp
@@ -0,0 +1,199 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioParam.h"
+#include "mozilla/dom/AudioParamBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioContext.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioParam)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioParam)
+  tmp->DisconnectFromGraphAndDestroyStream();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mNode)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(AudioParam)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_WRAPPERCACHE(AudioParam)
+
+NS_IMPL_CYCLE_COLLECTING_NATIVE_ADDREF(AudioParam)
+NS_IMPL_CYCLE_COLLECTING_NATIVE_RELEASE(AudioParam)
+
+NS_IMPL_CYCLE_COLLECTION_ROOT_NATIVE(AudioParam, AddRef)
+NS_IMPL_CYCLE_COLLECTION_UNROOT_NATIVE(AudioParam, Release)
+
+AudioParam::AudioParam(AudioNode* aNode,
+                       uint32_t aIndex,
+                       float aDefaultValue,
+                       const char* aName)
+  : AudioParamTimeline(aDefaultValue)
+  , mNode(aNode)
+  , mName(aName)
+  , mIndex(aIndex)
+  , mDefaultValue(aDefaultValue)
+{
+}
+
+AudioParam::~AudioParam()
+{
+  DisconnectFromGraphAndDestroyStream();
+}
+
+JSObject*
+AudioParam::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioParamBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+AudioParam::DisconnectFromGraphAndDestroyStream()
+{
+  MOZ_ASSERT(mRefCnt.get() > mInputNodes.Length(),
+             "Caller should be holding a reference or have called "
+             "mRefCnt.stabilizeForDeletion()");
+
+  while (!mInputNodes.IsEmpty()) {
+    uint32_t i = mInputNodes.Length() - 1;
+    RefPtr<AudioNode> input = mInputNodes[i].mInputNode;
+    mInputNodes.RemoveElementAt(i);
+    input->RemoveOutputParam(this);
+  }
+
+  if (mNodeStreamPort) {
+    mNodeStreamPort->Destroy();
+    mNodeStreamPort = nullptr;
+  }
+
+  if (mStream) {
+    mStream->Destroy();
+    mStream = nullptr;
+  }
+}
+
+MediaStream*
+AudioParam::Stream()
+{
+  if (mStream) {
+    return mStream;
+  }
+
+  AudioNodeEngine* engine = new AudioNodeEngine(nullptr);
+  RefPtr<AudioNodeStream> stream =
+    AudioNodeStream::Create(mNode->Context(), engine,
+                            AudioNodeStream::NO_STREAM_FLAGS,
+                            mNode->Context()->Graph());
+
+  // Force the input to have only one channel, and make it down-mix using
+  // the speaker rules if needed.
+  stream->SetChannelMixingParametersImpl(1, ChannelCountMode::Explicit, ChannelInterpretation::Speakers);
+  // Mark as an AudioParam helper stream
+  stream->SetAudioParamHelperStream();
+
+  mStream = stream.forget();
+
+  // Setup the AudioParam's stream as an input to the owner AudioNode's stream
+  AudioNodeStream* nodeStream = mNode->GetStream();
+  if (nodeStream) {
+    mNodeStreamPort =
+      nodeStream->AllocateInputPort(mStream, AudioNodeStream::AUDIO_TRACK);
+  }
+
+  // Send the stream to the timeline on the MSG side.
+  AudioTimelineEvent event(mStream);
+  SendEventToEngine(event);
+
+  return mStream;
+}
+
+static const char*
+ToString(AudioTimelineEvent::Type aType)
+{
+  switch (aType) {
+    case AudioTimelineEvent::SetValue:
+      return "SetValue";
+    case AudioTimelineEvent::SetValueAtTime:
+      return "SetValueAtTime";
+    case AudioTimelineEvent::LinearRamp:
+      return "LinearRamp";
+    case AudioTimelineEvent::ExponentialRamp:
+      return "ExponentialRamp";
+    case AudioTimelineEvent::SetTarget:
+      return "SetTarget";
+    case AudioTimelineEvent::SetValueCurve:
+      return "SetValueCurve";
+    case AudioTimelineEvent::Stream:
+      return "Stream";
+    case AudioTimelineEvent::Cancel:
+      return "Cancel";
+    default:
+      return "unknown AudioTimelineEvent";
+  }
+}
+
+void
+AudioParam::SendEventToEngine(const AudioTimelineEvent& aEvent)
+{
+  WEB_AUDIO_API_LOG("%f: %s for %u %s %s=%g time=%f %s=%g",
+                    GetParentObject()->CurrentTime(),
+                    mName, ParentNodeId(), ToString(aEvent.mType),
+                    aEvent.mType == AudioTimelineEvent::SetValueCurve ?
+                      "length" : "value",
+                    aEvent.mType == AudioTimelineEvent::SetValueCurve ?
+                      static_cast<double>(aEvent.mCurveLength) :
+                      static_cast<double>(aEvent.mValue),
+                    aEvent.Time<double>(),
+                    aEvent.mType == AudioTimelineEvent::SetValueCurve ?
+                      "duration" : "constant",
+                    aEvent.mType == AudioTimelineEvent::SetValueCurve ?
+                      aEvent.mDuration : aEvent.mTimeConstant);
+
+  AudioNodeStream* stream = mNode->GetStream();
+  if (stream) {
+    stream->SendTimelineEvent(mIndex, aEvent);
+  }
+}
+
+void
+AudioParam::CleanupOldEvents()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  double currentTime = mNode->Context()->CurrentTime();
+
+  CleanupEventsOlderThan(currentTime);
+}
+
+float
+AudioParamTimeline::AudioNodeInputValue(size_t aCounter) const
+{
+  MOZ_ASSERT(mStream);
+
+  // If we have a chunk produced by the AudioNode inputs to the AudioParam,
+  // get its value now.  We use aCounter to tell us which frame of the last
+  // AudioChunk to look at.
+  float audioNodeInputValue = 0.0f;
+  const AudioBlock& lastAudioNodeChunk =
+    static_cast<AudioNodeStream*>(mStream.get())->LastChunks()[0];
+  if (!lastAudioNodeChunk.IsNull()) {
+    MOZ_ASSERT(lastAudioNodeChunk.GetDuration() == WEBAUDIO_BLOCK_SIZE);
+    audioNodeInputValue =
+      static_cast<const float*>(lastAudioNodeChunk.mChannelData[0])[aCounter];
+    audioNodeInputValue *= lastAudioNodeChunk.mVolume;
+  }
+
+  return audioNodeInputValue;
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/AudioParam.h b/dom/media/webaudio/AudioParam.h
new file mode 100644
index 0000000000..90686cb898
--- /dev/null
+++ b/dom/media/webaudio/AudioParam.h
@@ -0,0 +1,246 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioParam_h_
+#define AudioParam_h_
+
+#include "AudioParamTimeline.h"
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "AudioNode.h"
+#include "mozilla/dom/TypedArray.h"
+#include "WebAudioUtils.h"
+#include "js/TypeDecls.h"
+
+namespace mozilla {
+
+namespace dom {
+
+class AudioParam final : public nsWrapperCache,
+                         public AudioParamTimeline
+{
+  virtual ~AudioParam();
+
+public:
+  AudioParam(AudioNode* aNode,
+             uint32_t aIndex,
+             float aDefaultValue,
+             const char* aName);
+
+  NS_IMETHOD_(MozExternalRefCountType) AddRef(void);
+  NS_IMETHOD_(MozExternalRefCountType) Release(void);
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(AudioParam)
+
+  AudioContext* GetParentObject() const
+  {
+    return mNode->Context();
+  }
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  // We override SetValueCurveAtTime to convert the Float32Array to the wrapper
+  // object.
+  AudioParam* SetValueCurveAtTime(const Float32Array& aValues,
+                                  double aStartTime,
+                                  double aDuration,
+                                  ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+    aValues.ComputeLengthAndData();
+
+    EventInsertionHelper(aRv, AudioTimelineEvent::SetValueCurve,
+                         aStartTime, 0.0f, 0.0f, aDuration, aValues.Data(),
+                         aValues.Length());
+    return this;
+  }
+
+  void SetValue(float aValue)
+  {
+    AudioTimelineEvent event(AudioTimelineEvent::SetValue, 0.0f, aValue);
+
+    ErrorResult rv;
+    if (!ValidateEvent(event, rv)) {
+      MOZ_ASSERT(false, "This should not happen, "
+                        "setting the value should always work");
+      return;
+    }
+
+    AudioParamTimeline::SetValue(aValue);
+
+    SendEventToEngine(event);
+  }
+
+  AudioParam* SetValueAtTime(float aValue, double aStartTime, ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+    EventInsertionHelper(aRv, AudioTimelineEvent::SetValueAtTime,
+                         aStartTime, aValue);
+
+    return this;
+  }
+
+  AudioParam* LinearRampToValueAtTime(float aValue, double aEndTime,
+                                      ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aEndTime)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+    EventInsertionHelper(aRv, AudioTimelineEvent::LinearRamp, aEndTime, aValue);
+    return this;
+  }
+
+  AudioParam* ExponentialRampToValueAtTime(float aValue, double aEndTime,
+                                           ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aEndTime)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+    EventInsertionHelper(aRv, AudioTimelineEvent::ExponentialRamp,
+                         aEndTime, aValue);
+    return this;
+  }
+
+  AudioParam* SetTargetAtTime(float aTarget, double aStartTime,
+                              double aTimeConstant, ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aStartTime) ||
+        !WebAudioUtils::IsTimeValid(aTimeConstant)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+    EventInsertionHelper(aRv, AudioTimelineEvent::SetTarget,
+                         aStartTime, aTarget,
+                         aTimeConstant);
+
+    return this;
+  }
+
+  AudioParam* CancelScheduledValues(double aStartTime, ErrorResult& aRv)
+  {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return this;
+    }
+
+    // Remove some events on the main thread copy.
+    AudioEventTimeline::CancelScheduledValues(aStartTime);
+
+    AudioTimelineEvent event(AudioTimelineEvent::Cancel, aStartTime, 0.0f);
+
+    SendEventToEngine(event);
+
+    return this;
+  }
+
+  uint32_t ParentNodeId()
+  {
+    return mNode->Id();
+  }
+
+  void GetName(nsAString& aName)
+  {
+    aName.AssignASCII(mName);
+  }
+
+  float DefaultValue() const
+  {
+    return mDefaultValue;
+  }
+
+  const nsTArray<AudioNode::InputNode>& InputNodes() const
+  {
+    return mInputNodes;
+  }
+
+  void RemoveInputNode(uint32_t aIndex)
+  {
+    mInputNodes.RemoveElementAt(aIndex);
+  }
+
+  AudioNode::InputNode* AppendInputNode()
+  {
+    return mInputNodes.AppendElement();
+  }
+
+  // May create the stream if it doesn't exist
+  MediaStream* Stream();
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioParamTimeline::SizeOfExcludingThis(aMallocSizeOf);
+    // Not owned:
+    // - mNode
+
+    // Just count the array, actual nodes are counted in mNode.
+    amount += mInputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+
+    if (mNodeStreamPort) {
+      amount += mNodeStreamPort->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  void EventInsertionHelper(ErrorResult& aRv,
+                            AudioTimelineEvent::Type aType,
+                            double aTime, float aValue,
+                            double aTimeConstant = 0.0,
+                            float aDuration = 0.0,
+                            const float* aCurve = nullptr,
+                            uint32_t aCurveLength = 0)
+  {
+    AudioTimelineEvent event(aType, aTime, aValue,
+                             aTimeConstant, aDuration, aCurve, aCurveLength);
+
+    if (!ValidateEvent(event, aRv)) {
+      return;
+    }
+
+    AudioEventTimeline::InsertEvent<double>(event);
+
+    SendEventToEngine(event);
+
+    CleanupOldEvents();
+  }
+
+  void CleanupOldEvents();
+
+  void SendEventToEngine(const AudioTimelineEvent& aEvent);
+
+  void DisconnectFromGraphAndDestroyStream();
+
+  nsCycleCollectingAutoRefCnt mRefCnt;
+  NS_DECL_OWNINGTHREAD
+  RefPtr<AudioNode> mNode;
+  // For every InputNode, there is a corresponding entry in mOutputParams of the
+  // InputNode's mInputNode.
+  nsTArray<AudioNode::InputNode> mInputNodes;
+  const char* mName;
+  // The input port used to connect the AudioParam's stream to its node's stream
+  RefPtr<MediaInputPort> mNodeStreamPort;
+  const uint32_t mIndex;
+  const float mDefaultValue;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/AudioParamTimeline.h b/dom/media/webaudio/AudioParamTimeline.h
new file mode 100644
index 0000000000..24ef753c3d
--- /dev/null
+++ b/dom/media/webaudio/AudioParamTimeline.h
@@ -0,0 +1,157 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioParamTimeline_h_
+#define AudioParamTimeline_h_
+
+#include "AudioEventTimeline.h"
+#include "mozilla/ErrorResult.h"
+#include "MediaStreamGraph.h"
+#include "AudioSegment.h"
+
+namespace mozilla {
+
+namespace dom {
+
+// This helper class is used to represent the part of the AudioParam
+// class that gets sent to AudioNodeEngine instances.  In addition to
+// AudioEventTimeline methods, it holds a pointer to an optional
+// MediaStream which represents the AudioNode inputs to the AudioParam.
+// This MediaStream is managed by the AudioParam subclass on the main
+// thread, and can only be obtained from the AudioNodeEngine instances
+// consuming this class.
+class AudioParamTimeline : public AudioEventTimeline
+{
+  typedef AudioEventTimeline BaseClass;
+
+public:
+  explicit AudioParamTimeline(float aDefaultValue)
+    : BaseClass(aDefaultValue)
+  {
+  }
+
+  MediaStream* Stream() const
+  {
+    return mStream;
+  }
+
+  bool HasSimpleValue() const
+  {
+    return BaseClass::HasSimpleValue() && !mStream;
+  }
+
+  template<class TimeType>
+  float GetValueAtTime(TimeType aTime)
+  {
+    return GetValueAtTime(aTime, 0);
+  }
+
+  template<typename TimeType>
+  void InsertEvent(const AudioTimelineEvent& aEvent)
+  {
+    if (aEvent.mType == AudioTimelineEvent::Cancel) {
+      CancelScheduledValues(aEvent.template Time<TimeType>());
+      return;
+    }
+    if (aEvent.mType == AudioTimelineEvent::Stream) {
+      mStream = aEvent.mStream;
+      return;
+    }
+    if (aEvent.mType == AudioTimelineEvent::SetValue) {
+      AudioEventTimeline::SetValue(aEvent.mValue);
+      return;
+    }
+    AudioEventTimeline::InsertEvent<TimeType>(aEvent);
+  }
+
+  // Get the value of the AudioParam at time aTime + aCounter.
+  // aCounter here is an offset to aTime if we try to get the value in ticks,
+  // otherwise it should always be zero.  aCounter is meant to be used when
+  template<class TimeType>
+  float GetValueAtTime(TimeType aTime, size_t aCounter);
+
+  // Get the values of the AudioParam at time aTime + (0 to aSize).
+  // aBuffer must have the correct aSize.
+  // aSize here is an offset to aTime if we try to get the value in ticks,
+  // otherwise it should always be zero.  aSize is meant to be used when
+  // getting the value of an a-rate AudioParam for each tick inside an
+  // AudioNodeEngine implementation.
+  template<class TimeType>
+  void GetValuesAtTime(TimeType aTime, float* aBuffer, const size_t aSize);
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    return mStream ? mStream->SizeOfIncludingThis(aMallocSizeOf) : 0;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+
+private:
+  float AudioNodeInputValue(size_t aCounter) const;
+
+protected:
+  // This is created lazily when needed.
+  RefPtr<MediaStream> mStream;
+};
+
+template<> inline float
+AudioParamTimeline::GetValueAtTime(double aTime, size_t aCounter)
+{
+  MOZ_ASSERT(!aCounter);
+
+  // Getting an AudioParam value on an AudioNode does not consider input from
+  // other AudioNodes, which is managed only on the graph thread.
+  return BaseClass::GetValueAtTime(aTime);
+}
+
+template<> inline float
+AudioParamTimeline::GetValueAtTime(int64_t aTime, size_t aCounter)
+{
+  MOZ_ASSERT(aCounter < WEBAUDIO_BLOCK_SIZE);
+  MOZ_ASSERT(!aCounter || !HasSimpleValue());
+
+  // Mix the value of the AudioParam itself with that of the AudioNode inputs.
+  return BaseClass::GetValueAtTime(static_cast<int64_t>(aTime + aCounter)) +
+    (mStream ? AudioNodeInputValue(aCounter) : 0.0f);
+}
+
+template<> inline void
+AudioParamTimeline::GetValuesAtTime(double aTime, float* aBuffer,
+                                    const size_t aSize)
+{
+  MOZ_ASSERT(aBuffer);
+  MOZ_ASSERT(aSize == 1);
+
+  // Getting an AudioParam value on an AudioNode does not consider input from
+  // other AudioNodes, which is managed only on the graph thread.
+  *aBuffer = BaseClass::GetValueAtTime(aTime);
+}
+
+template<> inline void
+AudioParamTimeline::GetValuesAtTime(int64_t aTime, float* aBuffer,
+                                    const size_t aSize)
+{
+  MOZ_ASSERT(aBuffer);
+  MOZ_ASSERT(aSize <= WEBAUDIO_BLOCK_SIZE);
+  MOZ_ASSERT(aSize == 1 || !HasSimpleValue());
+
+  // Mix the value of the AudioParam itself with that of the AudioNode inputs.
+  BaseClass::GetValuesAtTime(aTime, aBuffer, aSize);
+  if (mStream) {
+    for (size_t i = 0; i < aSize; ++i) {
+      aBuffer[i] += AudioNodeInputValue(i);
+    }
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioProcessingEvent.cpp b/dom/media/webaudio/AudioProcessingEvent.cpp
new file mode 100644
index 0000000000..01b9585caa
--- /dev/null
+++ b/dom/media/webaudio/AudioProcessingEvent.cpp
@@ -0,0 +1,57 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioProcessingEvent.h"
+#include "mozilla/dom/AudioProcessingEventBinding.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "AudioContext.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioProcessingEvent, Event,
+                                   mInputBuffer, mOutputBuffer, mNode)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioProcessingEvent)
+NS_INTERFACE_MAP_END_INHERITING(Event)
+
+NS_IMPL_ADDREF_INHERITED(AudioProcessingEvent, Event)
+NS_IMPL_RELEASE_INHERITED(AudioProcessingEvent, Event)
+
+AudioProcessingEvent::AudioProcessingEvent(ScriptProcessorNode* aOwner,
+                                           nsPresContext* aPresContext,
+                                           WidgetEvent* aEvent)
+  : Event(aOwner, aPresContext, aEvent)
+  , mPlaybackTime(0.0)
+  , mNode(aOwner)
+{
+}
+
+AudioProcessingEvent::~AudioProcessingEvent()
+{
+}
+
+JSObject*
+AudioProcessingEvent::WrapObjectInternal(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return AudioProcessingEventBinding::Wrap(aCx, this, aGivenProto);
+}
+
+already_AddRefed<AudioBuffer>
+AudioProcessingEvent::LazilyCreateBuffer(uint32_t aNumberOfChannels,
+                                         ErrorResult& aRv)
+{
+  RefPtr<AudioBuffer> buffer =
+    AudioBuffer::Create(mNode->Context(), aNumberOfChannels,
+                        mNode->BufferSize(),
+                        mNode->Context()->SampleRate(), aRv);
+  MOZ_ASSERT(buffer || aRv.ErrorCodeIs(NS_ERROR_OUT_OF_MEMORY));
+  return buffer.forget();
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/AudioProcessingEvent.h b/dom/media/webaudio/AudioProcessingEvent.h
new file mode 100644
index 0000000000..7b3b54d3ed
--- /dev/null
+++ b/dom/media/webaudio/AudioProcessingEvent.h
@@ -0,0 +1,85 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioProcessingEvent_h_
+#define AudioProcessingEvent_h_
+
+#include "AudioBuffer.h"
+#include "ScriptProcessorNode.h"
+#include "mozilla/dom/Event.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioProcessingEvent final : public Event
+{
+public:
+  AudioProcessingEvent(ScriptProcessorNode* aOwner,
+                       nsPresContext* aPresContext,
+                       WidgetEvent* aEvent);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_FORWARD_TO_EVENT
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioProcessingEvent, Event)
+
+  JSObject* WrapObjectInternal(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void InitEvent(AudioBuffer* aInputBuffer,
+                 uint32_t aNumberOfInputChannels,
+                 double aPlaybackTime)
+  {
+    InitEvent(NS_LITERAL_STRING("audioprocess"), false, false);
+    mInputBuffer = aInputBuffer;
+    mNumberOfInputChannels = aNumberOfInputChannels;
+    mPlaybackTime = aPlaybackTime;
+  }
+
+  double PlaybackTime() const
+  {
+    return mPlaybackTime;
+  }
+
+  AudioBuffer* GetInputBuffer(ErrorResult& aRv)
+  {
+    if (!mInputBuffer) {
+      mInputBuffer = LazilyCreateBuffer(mNumberOfInputChannels, aRv);
+    }
+    return mInputBuffer;
+  }
+
+  AudioBuffer* GetOutputBuffer(ErrorResult& aRv)
+  {
+    if (!mOutputBuffer) {
+      mOutputBuffer = LazilyCreateBuffer(mNode->NumberOfOutputChannels(), aRv);
+    }
+    return mOutputBuffer;
+  }
+
+  bool HasOutputBuffer() const
+  {
+    return !!mOutputBuffer;
+  }
+
+protected:
+  virtual ~AudioProcessingEvent();
+
+private:
+  already_AddRefed<AudioBuffer>
+  LazilyCreateBuffer(uint32_t aNumberOfChannels, ErrorResult& rv);
+
+private:
+  double mPlaybackTime;
+  RefPtr<AudioBuffer> mInputBuffer;
+  RefPtr<AudioBuffer> mOutputBuffer;
+  RefPtr<ScriptProcessorNode> mNode;
+  uint32_t mNumberOfInputChannels;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/BiquadFilterNode.cpp b/dom/media/webaudio/BiquadFilterNode.cpp
new file mode 100644
index 0000000000..0c8c05586b
--- /dev/null
+++ b/dom/media/webaudio/BiquadFilterNode.cpp
@@ -0,0 +1,355 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "BiquadFilterNode.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "PlayingRefChangeHandler.h"
+#include "WebAudioUtils.h"
+#include "blink/Biquad.h"
+#include "mozilla/UniquePtr.h"
+#include "AudioParamTimeline.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(BiquadFilterNode, AudioNode,
+                                   mFrequency, mDetune, mQ, mGain)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(BiquadFilterNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(BiquadFilterNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(BiquadFilterNode, AudioNode)
+
+static void
+SetParamsOnBiquad(WebCore::Biquad& aBiquad,
+                  float aSampleRate,
+                  BiquadFilterType aType,
+                  double aFrequency,
+                  double aQ,
+                  double aGain,
+                  double aDetune)
+{
+  const double nyquist = aSampleRate * 0.5;
+  double normalizedFrequency = aFrequency / nyquist;
+
+  if (aDetune) {
+    normalizedFrequency *= std::pow(2.0, aDetune / 1200);
+  }
+
+  switch (aType) {
+  case BiquadFilterType::Lowpass:
+    aBiquad.setLowpassParams(normalizedFrequency, aQ);
+    break;
+  case BiquadFilterType::Highpass:
+    aBiquad.setHighpassParams(normalizedFrequency, aQ);
+    break;
+  case BiquadFilterType::Bandpass:
+    aBiquad.setBandpassParams(normalizedFrequency, aQ);
+    break;
+  case BiquadFilterType::Lowshelf:
+    aBiquad.setLowShelfParams(normalizedFrequency, aGain);
+    break;
+  case BiquadFilterType::Highshelf:
+    aBiquad.setHighShelfParams(normalizedFrequency, aGain);
+    break;
+  case BiquadFilterType::Peaking:
+    aBiquad.setPeakingParams(normalizedFrequency, aQ, aGain);
+    break;
+  case BiquadFilterType::Notch:
+    aBiquad.setNotchParams(normalizedFrequency, aQ);
+    break;
+  case BiquadFilterType::Allpass:
+    aBiquad.setAllpassParams(normalizedFrequency, aQ);
+    break;
+  default:
+    NS_NOTREACHED("We should never see the alternate names here");
+    break;
+  }
+}
+
+class BiquadFilterNodeEngine final : public AudioNodeEngine
+{
+public:
+  BiquadFilterNodeEngine(AudioNode* aNode,
+                         AudioDestinationNode* aDestination,
+                         uint64_t aWindowID)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default values in sync with the default values in
+    // BiquadFilterNode::BiquadFilterNode
+    , mType(BiquadFilterType::Lowpass)
+    , mFrequency(350.f)
+    , mDetune(0.f)
+    , mQ(1.f)
+    , mGain(0.f)
+    , mWindowID(aWindowID)
+  {
+  }
+
+  enum Parameteres {
+    TYPE,
+    FREQUENCY,
+    DETUNE,
+    Q,
+    GAIN
+  };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override
+  {
+    switch (aIndex) {
+    case TYPE: mType = static_cast<BiquadFilterType>(aValue); break;
+    default:
+      NS_ERROR("Bad BiquadFilterNode Int32Parameter");
+    }
+  }
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case FREQUENCY:
+      mFrequency.InsertEvent<int64_t>(aEvent);
+      break;
+    case DETUNE:
+      mDetune.InsertEvent<int64_t>(aEvent);
+      break;
+    case Q:
+      mQ.InsertEvent<int64_t>(aEvent);
+      break;
+    case GAIN:
+      mGain.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad BiquadFilterNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    float inputBuffer[WEBAUDIO_BLOCK_SIZE + 4];
+    float* alignedInputBuffer = ALIGNED16(inputBuffer);
+    ASSERT_ALIGNED16(alignedInputBuffer);
+
+    if (aInput.IsNull()) {
+      bool hasTail = false;
+      for (uint32_t i = 0; i < mBiquads.Length(); ++i) {
+        if (mBiquads[i].hasTail()) {
+          hasTail = true;
+          break;
+        }
+      }
+      if (!hasTail) {
+        if (!mBiquads.IsEmpty()) {
+          mBiquads.Clear();
+          aStream->ScheduleCheckForInactive();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::RELEASE);
+          aStream->Graph()->
+            DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+        }
+
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+      }
+
+      PodArrayZero(inputBuffer);
+
+    } else if(mBiquads.Length() != aInput.ChannelCount()){
+      if (mBiquads.IsEmpty()) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+          new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      } else { // Help people diagnose bug 924718
+        WebAudioUtils::LogToDeveloperConsole(mWindowID,
+                                             "BiquadFilterChannelCountChangeWarning");
+      }
+
+      // Adjust the number of biquads based on the number of channels
+      mBiquads.SetLength(aInput.ChannelCount());
+    }
+
+    uint32_t numberOfChannels = mBiquads.Length();
+    aOutput->AllocateChannels(numberOfChannels);
+
+    StreamTime pos = mDestination->GraphTimeToStreamTime(aFrom);
+
+    double freq = mFrequency.GetValueAtTime(pos);
+    double q = mQ.GetValueAtTime(pos);
+    double gain = mGain.GetValueAtTime(pos);
+    double detune = mDetune.GetValueAtTime(pos);
+
+    for (uint32_t i = 0; i < numberOfChannels; ++i) {
+      const float* input;
+      if (aInput.IsNull()) {
+        input = alignedInputBuffer;
+      } else {
+        input = static_cast<const float*>(aInput.mChannelData[i]);
+        if (aInput.mVolume != 1.0) {
+          AudioBlockCopyChannelWithScale(input, aInput.mVolume, alignedInputBuffer);
+          input = alignedInputBuffer;
+        }
+      }
+      SetParamsOnBiquad(mBiquads[i], aStream->SampleRate(), mType, freq, q, gain, detune);
+
+      mBiquads[i].process(input,
+                          aOutput->ChannelFloatsForWrite(i),
+                          aInput.GetDuration());
+    }
+  }
+
+  bool IsActive() const override
+  {
+    return !mBiquads.IsEmpty();
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mDestination - probably not owned
+    // - AudioParamTimelines - counted in the AudioNode
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mBiquads.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  AudioNodeStream* mDestination;
+  BiquadFilterType mType;
+  AudioParamTimeline mFrequency;
+  AudioParamTimeline mDetune;
+  AudioParamTimeline mQ;
+  AudioParamTimeline mGain;
+  nsTArray<WebCore::Biquad> mBiquads;
+  uint64_t mWindowID;
+};
+
+BiquadFilterNode::BiquadFilterNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mType(BiquadFilterType::Lowpass)
+  , mFrequency(new AudioParam(this, BiquadFilterNodeEngine::FREQUENCY,
+                              350.f, "frequency"))
+  , mDetune(new AudioParam(this, BiquadFilterNodeEngine::DETUNE, 0.f, "detune"))
+  , mQ(new AudioParam(this, BiquadFilterNodeEngine::Q, 1.f, "Q"))
+  , mGain(new AudioParam(this, BiquadFilterNodeEngine::GAIN, 0.f, "gain"))
+{
+  uint64_t windowID = aContext->GetParentObject()->WindowID();
+  BiquadFilterNodeEngine* engine = new BiquadFilterNodeEngine(this, aContext->Destination(), windowID);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+BiquadFilterNode::~BiquadFilterNode()
+{
+}
+
+size_t
+BiquadFilterNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  if (mFrequency) {
+    amount += mFrequency->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mDetune) {
+    amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mQ) {
+    amount += mQ->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mGain) {
+    amount += mGain->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t
+BiquadFilterNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+BiquadFilterNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return BiquadFilterNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+BiquadFilterNode::SetType(BiquadFilterType aType)
+{
+  mType = aType;
+  SendInt32ParameterToStream(BiquadFilterNodeEngine::TYPE,
+                             static_cast<int32_t>(aType));
+}
+
+void
+BiquadFilterNode::GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                                       const Float32Array& aMagResponse,
+                                       const Float32Array& aPhaseResponse)
+{
+  aFrequencyHz.ComputeLengthAndData();
+  aMagResponse.ComputeLengthAndData();
+  aPhaseResponse.ComputeLengthAndData();
+
+  uint32_t length = std::min(std::min(aFrequencyHz.Length(), aMagResponse.Length()),
+                             aPhaseResponse.Length());
+  if (!length) {
+    return;
+  }
+
+  auto frequencies = MakeUnique<float[]>(length);
+  float* frequencyHz = aFrequencyHz.Data();
+  const double nyquist = Context()->SampleRate() * 0.5;
+
+  // Normalize the frequencies
+  for (uint32_t i = 0; i < length; ++i) {
+    if (frequencyHz[i] >= 0 && frequencyHz[i] <= nyquist) {
+        frequencies[i] = static_cast<float>(frequencyHz[i] / nyquist);
+    } else {
+        frequencies[i] = std::numeric_limits<float>::quiet_NaN();
+    }
+  }
+
+  const double currentTime = Context()->CurrentTime();
+
+  double freq = mFrequency->GetValueAtTime(currentTime);
+  double q = mQ->GetValueAtTime(currentTime);
+  double gain = mGain->GetValueAtTime(currentTime);
+  double detune = mDetune->GetValueAtTime(currentTime);
+
+  WebCore::Biquad biquad;
+  SetParamsOnBiquad(biquad, Context()->SampleRate(), mType, freq, q, gain, detune);
+  biquad.getFrequencyResponse(int(length), frequencies.get(), aMagResponse.Data(), aPhaseResponse.Data());
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/BiquadFilterNode.h b/dom/media/webaudio/BiquadFilterNode.h
new file mode 100644
index 0000000000..f81c623f0b
--- /dev/null
+++ b/dom/media/webaudio/BiquadFilterNode.h
@@ -0,0 +1,82 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef BiquadFilterNode_h_
+#define BiquadFilterNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/BiquadFilterNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class BiquadFilterNode final : public AudioNode
+{
+public:
+  explicit BiquadFilterNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(BiquadFilterNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  BiquadFilterType Type() const
+  {
+    return mType;
+  }
+  void SetType(BiquadFilterType aType);
+
+  AudioParam* Frequency() const
+  {
+    return mFrequency;
+  }
+
+  AudioParam* Detune() const
+  {
+    return mDetune;
+  }
+
+  AudioParam* Q() const
+  {
+    return mQ;
+  }
+
+  AudioParam* Gain() const
+  {
+    return mGain;
+  }
+
+  void GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                            const Float32Array& aMagResponse,
+                            const Float32Array& aPhaseResponse);
+
+  const char* NodeType() const override
+  {
+    return "BiquadFilterNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~BiquadFilterNode();
+
+private:
+  BiquadFilterType mType;
+  RefPtr<AudioParam> mFrequency;
+  RefPtr<AudioParam> mDetune;
+  RefPtr<AudioParam> mQ;
+  RefPtr<AudioParam> mGain;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/BufferDecoder.cpp b/dom/media/webaudio/BufferDecoder.cpp
new file mode 100644
index 0000000000..053a13becc
--- /dev/null
+++ b/dom/media/webaudio/BufferDecoder.cpp
@@ -0,0 +1,77 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "BufferDecoder.h"
+
+#include "nsISupports.h"
+#include "MediaResource.h"
+#include "GMPService.h"
+
+namespace mozilla {
+
+NS_IMPL_ISUPPORTS0(BufferDecoder)
+
+BufferDecoder::BufferDecoder(MediaResource* aResource, GMPCrashHelper* aCrashHelper)
+  : mResource(aResource)
+  , mCrashHelper(aCrashHelper)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_CTOR(BufferDecoder);
+}
+
+BufferDecoder::~BufferDecoder()
+{
+  // The dtor may run on any thread, we cannot be sure.
+  MOZ_COUNT_DTOR(BufferDecoder);
+}
+
+void
+BufferDecoder::BeginDecoding(TaskQueue* aTaskQueueIdentity)
+{
+  MOZ_ASSERT(!mTaskQueueIdentity && aTaskQueueIdentity);
+  mTaskQueueIdentity = aTaskQueueIdentity;
+}
+
+MediaResource*
+BufferDecoder::GetResource() const
+{
+  return mResource;
+}
+
+void
+BufferDecoder::NotifyDecodedFrames(const FrameStatisticsData& aStats)
+{
+  // ignore
+}
+
+VideoFrameContainer*
+BufferDecoder::GetVideoFrameContainer()
+{
+  // no video frame
+  return nullptr;
+}
+
+layers::ImageContainer*
+BufferDecoder::GetImageContainer()
+{
+  // no image container
+  return nullptr;
+}
+
+MediaDecoderOwner*
+BufferDecoder::GetOwner() const
+{
+  // unknown
+  return nullptr;
+}
+
+already_AddRefed<GMPCrashHelper>
+BufferDecoder::GetCrashHelper()
+{
+  return do_AddRef(mCrashHelper);
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/BufferDecoder.h b/dom/media/webaudio/BufferDecoder.h
new file mode 100644
index 0000000000..52cb92489d
--- /dev/null
+++ b/dom/media/webaudio/BufferDecoder.h
@@ -0,0 +1,54 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef BUFFER_DECODER_H_
+#define BUFFER_DECODER_H_
+
+#include "mozilla/Attributes.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/TaskQueue.h"
+
+#include "AbstractMediaDecoder.h"
+
+namespace mozilla {
+
+/**
+ * This class provides a decoder object which decodes a media file that lives in
+ * a memory buffer.
+ */
+class BufferDecoder final : public AbstractMediaDecoder
+{
+public:
+  // This class holds a weak pointer to MediaResource.  It's the responsibility
+  // of the caller to manage the memory of the MediaResource object.
+  explicit BufferDecoder(MediaResource* aResource, GMPCrashHelper* aCrashHelper);
+
+  NS_DECL_THREADSAFE_ISUPPORTS
+
+  // This has to be called before decoding begins
+  void BeginDecoding(TaskQueue* aTaskQueueIdentity);
+
+  MediaResource* GetResource() const final override;
+
+  void NotifyDecodedFrames(const FrameStatisticsData& aStats) final override;
+
+  VideoFrameContainer* GetVideoFrameContainer() final override;
+  layers::ImageContainer* GetImageContainer() final override;
+
+  MediaDecoderOwner* GetOwner() const final override;
+
+  already_AddRefed<GMPCrashHelper> GetCrashHelper() override;
+
+private:
+  virtual ~BufferDecoder();
+  RefPtr<TaskQueue> mTaskQueueIdentity;
+  RefPtr<MediaResource> mResource;
+  RefPtr<GMPCrashHelper> mCrashHelper;
+};
+
+} // namespace mozilla
+
+#endif /* BUFFER_DECODER_H_ */
diff --git a/dom/media/webaudio/ChannelMergerNode.cpp b/dom/media/webaudio/ChannelMergerNode.cpp
new file mode 100644
index 0000000000..7b63a98a60
--- /dev/null
+++ b/dom/media/webaudio/ChannelMergerNode.cpp
@@ -0,0 +1,90 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mozilla/dom/ChannelMergerNode.h"
+#include "mozilla/dom/ChannelMergerNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_ISUPPORTS_INHERITED0(ChannelMergerNode, AudioNode)
+
+class ChannelMergerNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit ChannelMergerNodeEngine(ChannelMergerNode* aNode)
+    : AudioNodeEngine(aNode)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void ProcessBlocksOnPorts(AudioNodeStream* aStream,
+                            const OutputChunks& aInput,
+                            OutputChunks& aOutput,
+                            bool* aFinished) override
+  {
+    MOZ_ASSERT(aInput.Length() >= 1, "Should have one or more input ports");
+
+    // Get the number of output channels, and allocate it
+    size_t channelCount = InputCount();
+    bool allNull = true;
+    for (size_t i = 0; i < channelCount; ++i) {
+      allNull &= aInput[i].IsNull();
+    }
+    if (allNull) {
+      aOutput[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    aOutput[0].AllocateChannels(channelCount);
+
+    for (size_t i = 0; i < channelCount; ++i) {
+      float* output = aOutput[0].ChannelFloatsForWrite(i);
+      if (aInput[i].IsNull()) {
+        PodZero(output, WEBAUDIO_BLOCK_SIZE);
+      } else {
+        AudioBlockCopyChannelWithScale(
+          static_cast<const float*>(aInput[i].mChannelData[0]),
+          aInput[i].mVolume, output);
+      }
+    }
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+};
+
+ChannelMergerNode::ChannelMergerNode(AudioContext* aContext,
+                                     uint16_t aInputCount)
+  : AudioNode(aContext,
+              1,
+              ChannelCountMode::Explicit,
+              ChannelInterpretation::Speakers)
+  , mInputCount(aInputCount)
+{
+  mStream = AudioNodeStream::Create(aContext,
+                                    new ChannelMergerNodeEngine(this),
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+ChannelMergerNode::~ChannelMergerNode()
+{
+}
+
+JSObject*
+ChannelMergerNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return ChannelMergerNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/ChannelMergerNode.h b/dom/media/webaudio/ChannelMergerNode.h
new file mode 100644
index 0000000000..d064c8e236
--- /dev/null
+++ b/dom/media/webaudio/ChannelMergerNode.h
@@ -0,0 +1,50 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ChannelMergerNode_h_
+#define ChannelMergerNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class ChannelMergerNode final : public AudioNode
+{
+public:
+  ChannelMergerNode(AudioContext* aContext,
+                    uint16_t aInputCount);
+
+  NS_DECL_ISUPPORTS_INHERITED
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfInputs() const override { return mInputCount; }
+
+  const char* NodeType() const override
+  {
+    return "ChannelMergerNode";
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+protected:
+  virtual ~ChannelMergerNode();
+
+private:
+  const uint16_t mInputCount;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/ChannelSplitterNode.cpp b/dom/media/webaudio/ChannelSplitterNode.cpp
new file mode 100644
index 0000000000..34a414d164
--- /dev/null
+++ b/dom/media/webaudio/ChannelSplitterNode.cpp
@@ -0,0 +1,81 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mozilla/dom/ChannelSplitterNode.h"
+#include "mozilla/dom/ChannelSplitterNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_ISUPPORTS_INHERITED0(ChannelSplitterNode, AudioNode)
+
+class ChannelSplitterNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit ChannelSplitterNodeEngine(ChannelSplitterNode* aNode)
+    : AudioNodeEngine(aNode)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void ProcessBlocksOnPorts(AudioNodeStream* aStream,
+                            const OutputChunks& aInput,
+                            OutputChunks& aOutput,
+                            bool* aFinished) override
+  {
+    MOZ_ASSERT(aInput.Length() == 1, "Should only have one input port");
+
+    aOutput.SetLength(OutputCount());
+    for (uint16_t i = 0; i < OutputCount(); ++i) {
+      if (i < aInput[0].ChannelCount()) {
+        // Split out existing channels
+        aOutput[i].AllocateChannels(1);
+        AudioBlockCopyChannelWithScale(
+            static_cast<const float*>(aInput[0].mChannelData[i]),
+            aInput[0].mVolume,
+            aOutput[i].ChannelFloatsForWrite(0));
+      } else {
+        // Pad with silent channels if needed
+        aOutput[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+};
+
+ChannelSplitterNode::ChannelSplitterNode(AudioContext* aContext,
+                                         uint16_t aOutputCount)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mOutputCount(aOutputCount)
+{
+  mStream = AudioNodeStream::Create(aContext,
+                                    new ChannelSplitterNodeEngine(this),
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+ChannelSplitterNode::~ChannelSplitterNode()
+{
+}
+
+JSObject*
+ChannelSplitterNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return ChannelSplitterNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/ChannelSplitterNode.h b/dom/media/webaudio/ChannelSplitterNode.h
new file mode 100644
index 0000000000..3b267eccc5
--- /dev/null
+++ b/dom/media/webaudio/ChannelSplitterNode.h
@@ -0,0 +1,50 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ChannelSplitterNode_h_
+#define ChannelSplitterNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class ChannelSplitterNode final : public AudioNode
+{
+public:
+  ChannelSplitterNode(AudioContext* aContext,
+                      uint16_t aOutputCount);
+
+  NS_DECL_ISUPPORTS_INHERITED
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const override { return mOutputCount; }
+
+  const char* NodeType() const override
+  {
+    return "ChannelSplitterNode";
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+protected:
+  virtual ~ChannelSplitterNode();
+
+private:
+  const uint16_t mOutputCount;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/ConstantSourceNode.cpp b/dom/media/webaudio/ConstantSourceNode.cpp
new file mode 100644
index 0000000000..b6884105cf
--- /dev/null
+++ b/dom/media/webaudio/ConstantSourceNode.cpp
@@ -0,0 +1,286 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "ConstantSourceNode.h"
+
+#include "AudioDestinationNode.h"
+#include "nsContentUtils.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(ConstantSourceNode, AudioNode,
+                                   mOffset)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(ConstantSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(ConstantSourceNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(ConstantSourceNode, AudioNode)
+
+class ConstantSourceNodeEngine final : public AudioNodeEngine
+{
+public:
+  ConstantSourceNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mSource(nullptr)
+    , mDestination(aDestination->Stream())
+    , mStart(-1)
+    , mStop(STREAM_TIME_MAX)
+    // Keep the default values in sync with ConstantSourceNode::ConstantSourceNode.
+    , mOffset(1.0f)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void SetSourceStream(AudioNodeStream* aSource)
+  {
+    mSource = aSource;
+  }
+
+  enum Parameters {
+    OFFSET,
+    START,
+    STOP,
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case OFFSET:
+      mOffset.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad ConstantSourceNodeEngine TimelineParameter");
+    }
+  }
+
+  void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) override
+  {
+    switch (aIndex) {
+    case START:
+      mStart = aParam;
+      mSource->SetActive();
+      break;
+    case STOP: mStop = aParam; break;
+    default:
+      NS_ERROR("Bad ConstantSourceNodeEngine StreamTimeParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    MOZ_ASSERT(mSource == aStream, "Invalid source stream");
+
+    StreamTime ticks = mDestination->GraphTimeToStreamTime(aFrom);
+    if (mStart == -1) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE <= mStart || ticks >= mStop) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else {
+      aOutput->AllocateChannels(1);
+      float* output = aOutput->ChannelFloatsForWrite(0);
+
+      if (mOffset.HasSimpleValue()) {
+        for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+          output[i] = mOffset.GetValueAtTime(aFrom, 0);
+        }
+      } else {
+        mOffset.GetValuesAtTime(ticks, output, WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE >= mStop) {
+      // We've finished playing.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // start() has been called.
+    return mStart != -1;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    // Not owned:
+    // - mSource
+    // - mDestination
+    // - mOffset (internal ref owned by node)
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mSource;
+  AudioNodeStream* mDestination;
+  StreamTime mStart;
+  StreamTime mStop;
+  AudioParamTimeline mOffset;
+};
+
+ConstantSourceNode::ConstantSourceNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              1,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mOffset(new AudioParam(this, ConstantSourceNodeEngine::OFFSET,
+                           1.0, "offset"))
+  , mStartCalled(false)
+{
+  ConstantSourceNodeEngine* engine = new ConstantSourceNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NEED_MAIN_THREAD_FINISHED,
+                                    aContext->Graph());
+  engine->SetSourceStream(mStream);
+  mStream->AddMainThreadListener(this);
+}
+
+ConstantSourceNode::~ConstantSourceNode()
+{
+}
+
+size_t
+ConstantSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  amount += mOffset->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+ConstantSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+ConstantSourceNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return ConstantSourceNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+already_AddRefed<ConstantSourceNode>
+ConstantSourceNode::Constructor(const GlobalObject& aGlobal,
+                                AudioContext& aContext,
+                                const ConstantSourceOptions& aOptions,
+                                ErrorResult& aRv)
+{
+  RefPtr<ConstantSourceNode> object = new ConstantSourceNode(&aContext);
+  object->mOffset->SetValue(aOptions.mOffset);
+  return object.forget();
+}
+
+void
+ConstantSourceNode::DestroyMediaStream()
+{
+  if (mStream) {
+    mStream->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaStream();
+}
+
+void
+ConstantSourceNode::Start(double aWhen, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+  mStartCalled = true;
+
+  if (!mStream) {
+    return;
+  }
+
+  mStream->SetStreamTimeParameter(ConstantSourceNodeEngine::START,
+                                  Context(), aWhen);
+
+  MarkActive();
+}
+
+void
+ConstantSourceNode::Stop(double aWhen, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+
+  if (!mStream || !Context()) {
+    return;
+  }
+
+  mStream->SetStreamTimeParameter(ConstantSourceNodeEngine::STOP,
+                                  Context(), std::max(0.0, aWhen));
+}
+
+void
+ConstantSourceNode::NotifyMainThreadStreamFinished()
+{
+  MOZ_ASSERT(mStream->IsFinished());
+
+  class EndedEventDispatcher final : public Runnable
+  {
+  public:
+    explicit EndedEventDispatcher(ConstantSourceNode* aNode)
+      : mNode(aNode) {}
+    NS_IMETHOD Run() override
+    {
+      // If it's not safe to run scripts right now, schedule this to run later
+      if (!nsContentUtils::IsSafeToRunScript()) {
+        nsContentUtils::AddScriptRunner(this);
+        return NS_OK;
+      }
+
+      mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
+      // Release stream resources.
+      mNode->DestroyMediaStream();
+      return NS_OK;
+    }
+  private:
+    RefPtr<ConstantSourceNode> mNode;
+  };
+
+  NS_DispatchToMainThread(new EndedEventDispatcher(this));
+
+  // Drop the playing reference
+  // Warning: The below line might delete this.
+  MarkInactive();
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/ConstantSourceNode.h b/dom/media/webaudio/ConstantSourceNode.h
new file mode 100644
index 0000000000..7b5e7197e9
--- /dev/null
+++ b/dom/media/webaudio/ConstantSourceNode.h
@@ -0,0 +1,76 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ConstantSourceNode_h_
+#define ConstantSourceNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/ConstantSourceNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class ConstantSourceNode final : public AudioNode,
+                                 public MainThreadMediaStreamListener
+{
+public:
+  explicit ConstantSourceNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ConstantSourceNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  static already_AddRefed<ConstantSourceNode>
+  Constructor(const GlobalObject& aGlobal,
+              AudioContext& aContext,
+              const ConstantSourceOptions& aOptions,
+              ErrorResult& aRv);
+
+  void DestroyMediaStream() override;
+
+  uint16_t NumberOfInputs() const final override
+  {
+    return 0;
+  }
+
+  AudioParam* Offset() const
+  {
+    return mOffset;
+  }
+
+  void Start(double aWhen, ErrorResult& rv);
+  void Stop(double aWhen, ErrorResult& rv);
+
+  IMPL_EVENT_HANDLER(ended)
+
+  void NotifyMainThreadStreamFinished() override;
+
+  const char* NodeType() const override
+  {
+    return "ConstantSourceNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~ConstantSourceNode();
+
+private:
+  RefPtr<AudioParam> mOffset;
+  bool mStartCalled;
+
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/ConvolverNode.cpp b/dom/media/webaudio/ConvolverNode.cpp
new file mode 100644
index 0000000000..314cdf7cf6
--- /dev/null
+++ b/dom/media/webaudio/ConvolverNode.cpp
@@ -0,0 +1,295 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "ConvolverNode.h"
+#include "mozilla/dom/ConvolverNodeBinding.h"
+#include "nsAutoPtr.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "blink/Reverb.h"
+#include "PlayingRefChangeHandler.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(ConvolverNode, AudioNode, mBuffer)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(ConvolverNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(ConvolverNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(ConvolverNode, AudioNode)
+
+class ConvolverNodeEngine final : public AudioNodeEngine
+{
+  typedef PlayingRefChangeHandler PlayingRefChanged;
+public:
+  ConvolverNodeEngine(AudioNode* aNode, bool aNormalize)
+    : AudioNodeEngine(aNode)
+    , mBufferLength(0)
+    , mLeftOverData(INT32_MIN)
+    , mSampleRate(0.0f)
+    , mUseBackgroundThreads(!aNode->Context()->IsOffline())
+    , mNormalize(aNormalize)
+  {
+  }
+
+  enum Parameters {
+    BUFFER_LENGTH,
+    SAMPLE_RATE,
+    NORMALIZE
+  };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    switch (aIndex) {
+    case BUFFER_LENGTH:
+      // BUFFER_LENGTH is the first parameter that we set when setting a new buffer,
+      // so we should be careful to invalidate the rest of our state here.
+      mBuffer = nullptr;
+      mSampleRate = 0.0f;
+      mBufferLength = aParam;
+      mLeftOverData = INT32_MIN;
+      break;
+    case SAMPLE_RATE:
+      mSampleRate = aParam;
+      break;
+    case NORMALIZE:
+      mNormalize = !!aParam;
+      break;
+    default:
+      NS_ERROR("Bad ConvolverNodeEngine Int32Parameter");
+    }
+  }
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override
+  {
+    switch (aIndex) {
+    case SAMPLE_RATE:
+      mSampleRate = aParam;
+      AdjustReverb();
+      break;
+    default:
+      NS_ERROR("Bad ConvolverNodeEngine DoubleParameter");
+    }
+  }
+  void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer) override
+  {
+    mBuffer = aBuffer;
+    AdjustReverb();
+  }
+
+  void AdjustReverb()
+  {
+    // Note about empirical tuning (this is copied from Blink)
+    // The maximum FFT size affects reverb performance and accuracy.
+    // If the reverb is single-threaded and processes entirely in the real-time audio thread,
+    // it's important not to make this too high.  In this case 8192 is a good value.
+    // But, the Reverb object is multi-threaded, so we want this as high as possible without losing too much accuracy.
+    // Very large FFTs will have worse phase errors. Given these constraints 32768 is a good compromise.
+    const size_t MaxFFTSize = 32768;
+
+    if (!mBuffer || !mBufferLength || !mSampleRate) {
+      mReverb = nullptr;
+      mLeftOverData = INT32_MIN;
+      return;
+    }
+
+    mReverb = new WebCore::Reverb(mBuffer, mBufferLength,
+                                  MaxFFTSize, 2, mUseBackgroundThreads,
+                                  mNormalize, mSampleRate);
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    if (!mReverb) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    AudioBlock input = aInput;
+    if (aInput.IsNull()) {
+      if (mLeftOverData > 0) {
+        mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+        input.AllocateChannels(1);
+        WriteZeroesToAudioBlock(&input, 0, WEBAUDIO_BLOCK_SIZE);
+      } else {
+        if (mLeftOverData != INT32_MIN) {
+          mLeftOverData = INT32_MIN;
+          aStream->ScheduleCheckForInactive();
+          RefPtr<PlayingRefChanged> refchanged =
+            new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE);
+          aStream->Graph()->
+            DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+        }
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+      }
+    } else {
+      if (aInput.mVolume != 1.0f) {
+        // Pre-multiply the input's volume
+        uint32_t numChannels = aInput.ChannelCount();
+        input.AllocateChannels(numChannels);
+        for (uint32_t i = 0; i < numChannels; ++i) {
+          const float* src = static_cast<const float*>(aInput.mChannelData[i]);
+          float* dest = input.ChannelFloatsForWrite(i);
+          AudioBlockCopyChannelWithScale(src, aInput.mVolume, dest);
+        }
+      }
+
+      if (mLeftOverData <= 0) {
+        RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      mLeftOverData = mBufferLength;
+      MOZ_ASSERT(mLeftOverData > 0);
+    }
+    aOutput->AllocateChannels(2);
+
+    mReverb->process(&input, aOutput);
+  }
+
+  bool IsActive() const override
+  {
+    return mLeftOverData != INT32_MIN;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    if (mBuffer && !mBuffer->IsShared()) {
+      amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    if (mReverb) {
+      amount += mReverb->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  nsAutoPtr<WebCore::Reverb> mReverb;
+  int32_t mBufferLength;
+  int32_t mLeftOverData;
+  float mSampleRate;
+  bool mUseBackgroundThreads;
+  bool mNormalize;
+};
+
+ConvolverNode::ConvolverNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Clamped_max,
+              ChannelInterpretation::Speakers)
+  , mNormalize(true)
+{
+  ConvolverNodeEngine* engine = new ConvolverNodeEngine(this, mNormalize);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+ConvolverNode::~ConvolverNode()
+{
+}
+
+size_t
+ConvolverNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  if (mBuffer) {
+    // NB: mBuffer might be shared with the associated engine, by convention
+    //     the AudioNode will report.
+    amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t
+ConvolverNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+ConvolverNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return ConvolverNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv)
+{
+  if (aBuffer) {
+    switch (aBuffer->NumberOfChannels()) {
+    case 1:
+    case 2:
+    case 4:
+      // Supported number of channels
+      break;
+    default:
+      aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
+      return;
+    }
+  }
+
+  mBuffer = aBuffer;
+
+  // Send the buffer to the stream
+  AudioNodeStream* ns = mStream;
+  MOZ_ASSERT(ns, "Why don't we have a stream here?");
+  if (mBuffer) {
+    uint32_t length = mBuffer->Length();
+    RefPtr<ThreadSharedFloatArrayBufferList> data =
+      mBuffer->GetThreadSharedChannelsForRate(aCx);
+    if (data && length < WEBAUDIO_BLOCK_SIZE) {
+      // For very small impulse response buffers, we need to pad the
+      // buffer with 0 to make sure that the Reverb implementation
+      // has enough data to compute FFTs from.
+      length = WEBAUDIO_BLOCK_SIZE;
+      RefPtr<ThreadSharedFloatArrayBufferList> paddedBuffer =
+        new ThreadSharedFloatArrayBufferList(data->GetChannels());
+      void* channelData = malloc(sizeof(float) * length * data->GetChannels() + 15);
+      float* alignedChannelData = ALIGNED16(channelData);
+      ASSERT_ALIGNED16(alignedChannelData);
+      for (uint32_t i = 0; i < data->GetChannels(); ++i) {
+        PodCopy(alignedChannelData + length * i, data->GetData(i), mBuffer->Length());
+        PodZero(alignedChannelData + length * i + mBuffer->Length(), WEBAUDIO_BLOCK_SIZE - mBuffer->Length());
+        paddedBuffer->SetData(i, (i == 0) ? channelData : nullptr, free, alignedChannelData);
+      }
+      data = paddedBuffer;
+    }
+    SendInt32ParameterToStream(ConvolverNodeEngine::BUFFER_LENGTH, length);
+    SendDoubleParameterToStream(ConvolverNodeEngine::SAMPLE_RATE,
+                                mBuffer->SampleRate());
+    ns->SetBuffer(data.forget());
+  } else {
+    ns->SetBuffer(nullptr);
+  }
+}
+
+void
+ConvolverNode::SetNormalize(bool aNormalize)
+{
+  mNormalize = aNormalize;
+  SendInt32ParameterToStream(ConvolverNodeEngine::NORMALIZE, aNormalize);
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/ConvolverNode.h b/dom/media/webaudio/ConvolverNode.h
new file mode 100644
index 0000000000..53cff9d273
--- /dev/null
+++ b/dom/media/webaudio/ConvolverNode.h
@@ -0,0 +1,78 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ConvolverNode_h_
+#define ConvolverNode_h_
+
+#include "AudioNode.h"
+#include "AudioBuffer.h"
+
+namespace mozilla {
+namespace dom {
+
+class ConvolverNode final : public AudioNode
+{
+public:
+  explicit ConvolverNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ConvolverNode, AudioNode);
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioBuffer* GetBuffer(JSContext* aCx) const
+  {
+    return mBuffer;
+  }
+
+  void SetBuffer(JSContext* aCx, AudioBuffer* aBufferi, ErrorResult& aRv);
+
+  bool Normalize() const
+  {
+    return mNormalize;
+  }
+
+  void SetNormalize(bool aNormal);
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override
+  {
+    if (aChannelCount > 2) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode, ErrorResult& aRv) override
+  {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  const char* NodeType() const override
+  {
+    return "ConvolverNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~ConvolverNode();
+
+private:
+  RefPtr<AudioBuffer> mBuffer;
+  bool mNormalize;
+};
+
+
+} //end namespace dom
+} //end namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/DelayBuffer.cpp b/dom/media/webaudio/DelayBuffer.cpp
new file mode 100644
index 0000000000..c7f7198c90
--- /dev/null
+++ b/dom/media/webaudio/DelayBuffer.cpp
@@ -0,0 +1,263 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "DelayBuffer.h"
+
+#include "mozilla/PodOperations.h"
+#include "AudioChannelFormat.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+
+size_t
+DelayBuffer::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = 0;
+  amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mChunks.Length(); i++) {
+    amount += mChunks[i].SizeOfExcludingThis(aMallocSizeOf, false);
+  }
+
+  amount += mUpmixChannels.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+void
+DelayBuffer::Write(const AudioBlock& aInputChunk)
+{
+  // We must have a reference to the buffer if there are channels
+  MOZ_ASSERT(aInputChunk.IsNull() == !aInputChunk.ChannelCount());
+#ifdef DEBUG
+  MOZ_ASSERT(!mHaveWrittenBlock);
+  mHaveWrittenBlock = true;
+#endif
+
+  if (!EnsureBuffer()) {
+    return;
+  }
+
+  if (mCurrentChunk == mLastReadChunk) {
+    mLastReadChunk = -1; // invalidate cache
+  }
+  mChunks[mCurrentChunk] = aInputChunk.AsAudioChunk();
+}
+
+void
+DelayBuffer::Read(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                  AudioBlock* aOutputChunk,
+                  ChannelInterpretation aChannelInterpretation)
+{
+  int chunkCount = mChunks.Length();
+  if (!chunkCount) {
+    aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  // Find the maximum number of contributing channels to determine the output
+  // channel count that retains all signal information.  Buffered blocks will
+  // be upmixed if necessary.
+  //
+  // First find the range of "delay" offsets backwards from the current
+  // position.  Note that these may be negative for frames that are after the
+  // current position (including i).
+  double minDelay = aPerFrameDelays[0];
+  double maxDelay = minDelay;
+  for (unsigned i = 1; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    minDelay = std::min(minDelay, aPerFrameDelays[i] - i);
+    maxDelay = std::max(maxDelay, aPerFrameDelays[i] - i);
+  }
+
+  // Now find the chunks touched by this range and check their channel counts.
+  int oldestChunk = ChunkForDelay(int(maxDelay) + 1);
+  int youngestChunk = ChunkForDelay(minDelay);
+
+  uint32_t channelCount = 0;
+  for (int i = oldestChunk; true; i = (i + 1) % chunkCount) {
+    channelCount = GetAudioChannelsSuperset(channelCount,
+                                            mChunks[i].ChannelCount());
+    if (i == youngestChunk) {
+      break;
+    }
+  }
+
+  if (channelCount) {
+    aOutputChunk->AllocateChannels(channelCount);
+    ReadChannels(aPerFrameDelays, aOutputChunk,
+                 0, channelCount, aChannelInterpretation);
+  } else {
+    aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+
+  // Remember currentDelayFrames for the next ProcessBlock call
+  mCurrentDelay = aPerFrameDelays[WEBAUDIO_BLOCK_SIZE - 1];
+}
+
+void
+DelayBuffer::ReadChannel(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                         AudioBlock* aOutputChunk, uint32_t aChannel,
+                         ChannelInterpretation aChannelInterpretation)
+{
+  if (!mChunks.Length()) {
+    float* outputChannel = aOutputChunk->ChannelFloatsForWrite(aChannel);
+    PodZero(outputChannel, WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  ReadChannels(aPerFrameDelays, aOutputChunk,
+               aChannel, 1, aChannelInterpretation);
+}
+
+void
+DelayBuffer::ReadChannels(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                          AudioBlock* aOutputChunk,
+                          uint32_t aFirstChannel, uint32_t aNumChannelsToRead,
+                          ChannelInterpretation aChannelInterpretation)
+{
+  uint32_t totalChannelCount = aOutputChunk->ChannelCount();
+  uint32_t readChannelsEnd = aFirstChannel + aNumChannelsToRead;
+  MOZ_ASSERT(readChannelsEnd <= totalChannelCount);
+
+  if (mUpmixChannels.Length() != totalChannelCount) {
+    mLastReadChunk = -1; // invalidate cache
+  }
+
+  for (uint32_t channel = aFirstChannel;
+       channel < readChannelsEnd; ++channel) {
+    PodZero(aOutputChunk->ChannelFloatsForWrite(channel), WEBAUDIO_BLOCK_SIZE);
+  }
+
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    double currentDelay = aPerFrameDelays[i];
+    MOZ_ASSERT(currentDelay >= 0.0);
+    MOZ_ASSERT(currentDelay <= (mChunks.Length() - 1) * WEBAUDIO_BLOCK_SIZE);
+
+    // Interpolate two input frames in case the read position does not match
+    // an integer index.
+    // Use the larger delay, for the older frame, first, as this is more
+    // likely to use the cached upmixed channel arrays.
+    int floorDelay = int(currentDelay);
+    double interpolationFactor = currentDelay - floorDelay;
+    int positions[2];
+    positions[1] = PositionForDelay(floorDelay) + i;
+    positions[0] = positions[1] - 1;
+
+    for (unsigned tick = 0; tick < ArrayLength(positions); ++tick) {
+      int readChunk = ChunkForPosition(positions[tick]);
+      // mVolume is not set on default initialized chunks so handle null
+      // chunks specially.
+      if (!mChunks[readChunk].IsNull()) {
+        int readOffset = OffsetForPosition(positions[tick]);
+        UpdateUpmixChannels(readChunk, totalChannelCount,
+                            aChannelInterpretation);
+        double multiplier = interpolationFactor * mChunks[readChunk].mVolume;
+        for (uint32_t channel = aFirstChannel;
+             channel < readChannelsEnd; ++channel) {
+          aOutputChunk->ChannelFloatsForWrite(channel)[i] += multiplier *
+            mUpmixChannels[channel][readOffset];
+        }
+      }
+
+      interpolationFactor = 1.0 - interpolationFactor;
+    }
+  }
+}
+
+void
+DelayBuffer::Read(double aDelayTicks, AudioBlock* aOutputChunk,
+                  ChannelInterpretation aChannelInterpretation)
+{
+  const bool firstTime = mCurrentDelay < 0.0;
+  double currentDelay = firstTime ? aDelayTicks : mCurrentDelay;
+
+  double computedDelay[WEBAUDIO_BLOCK_SIZE];
+
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    // If the value has changed, smoothly approach it
+    currentDelay += (aDelayTicks - currentDelay) * mSmoothingRate;
+    computedDelay[i] = currentDelay;
+  }
+
+  Read(computedDelay, aOutputChunk, aChannelInterpretation);
+}
+
+bool
+DelayBuffer::EnsureBuffer()
+{
+  if (mChunks.Length() == 0) {
+    // The length of the buffer is at least one block greater than the maximum
+    // delay so that writing an input block does not overwrite the block that
+    // would subsequently be read at maximum delay.  Also round up to the next
+    // block size, so that no block of writes will need to wrap.
+    const int chunkCount = (mMaxDelayTicks + 2 * WEBAUDIO_BLOCK_SIZE - 1) >>
+                                         WEBAUDIO_BLOCK_SIZE_BITS;
+    if (!mChunks.SetLength(chunkCount, fallible)) {
+      return false;
+    }
+
+    mLastReadChunk = -1;
+  }
+  return true;
+}
+
+int
+DelayBuffer::PositionForDelay(int aDelay) {
+  // Adding mChunks.Length() keeps integers positive for defined and
+  // appropriate bitshift, remainder, and bitwise operations.
+  return ((mCurrentChunk + mChunks.Length()) * WEBAUDIO_BLOCK_SIZE) - aDelay;
+}
+
+int
+DelayBuffer::ChunkForPosition(int aPosition)
+{
+  MOZ_ASSERT(aPosition >= 0);
+  return (aPosition >> WEBAUDIO_BLOCK_SIZE_BITS) % mChunks.Length();
+}
+
+int
+DelayBuffer::OffsetForPosition(int aPosition)
+{
+  MOZ_ASSERT(aPosition >= 0);
+  return aPosition & (WEBAUDIO_BLOCK_SIZE - 1);
+}
+
+int
+DelayBuffer::ChunkForDelay(int aDelay)
+{
+  return ChunkForPosition(PositionForDelay(aDelay));
+}
+
+void
+DelayBuffer::UpdateUpmixChannels(int aNewReadChunk, uint32_t aChannelCount,
+                                 ChannelInterpretation aChannelInterpretation)
+{
+  if (aNewReadChunk == mLastReadChunk) {
+    MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount);
+    return;
+  }
+
+  NS_WARNING_ASSERTION(mHaveWrittenBlock || aNewReadChunk != mCurrentChunk,
+                       "Smoothing is making feedback delay too small.");
+
+  mLastReadChunk = aNewReadChunk;
+  mUpmixChannels = mChunks[aNewReadChunk].ChannelData<float>();
+  MOZ_ASSERT(mUpmixChannels.Length() <= aChannelCount);
+  if (mUpmixChannels.Length() < aChannelCount) {
+    if (aChannelInterpretation == ChannelInterpretation::Speakers) {
+      AudioChannelsUpMix(&mUpmixChannels,
+                         aChannelCount, SilentChannel::ZeroChannel<float>());
+      MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount,
+                 "We called GetAudioChannelsSuperset to avoid this");
+    } else {
+      // Fill up the remaining channels with zeros
+      for (uint32_t channel = mUpmixChannels.Length();
+           channel < aChannelCount; ++channel) {
+        mUpmixChannels.AppendElement(SilentChannel::ZeroChannel<float>());
+      }
+    }
+  }
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/DelayBuffer.h b/dom/media/webaudio/DelayBuffer.h
new file mode 100644
index 0000000000..e55d0ba83a
--- /dev/null
+++ b/dom/media/webaudio/DelayBuffer.h
@@ -0,0 +1,115 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DelayBuffer_h_
+#define DelayBuffer_h_
+
+#include "nsTArray.h"
+#include "AudioSegment.h"
+#include "mozilla/dom/AudioNodeBinding.h" // for ChannelInterpretation
+
+namespace mozilla {
+
+class DelayBuffer final
+{
+  typedef dom::ChannelInterpretation ChannelInterpretation;
+
+public:
+  // See WebAudioUtils::ComputeSmoothingRate() for frame to frame exponential
+  // |smoothingRate| multiplier.
+  DelayBuffer(double aMaxDelayTicks, double aSmoothingRate)
+    : mSmoothingRate(aSmoothingRate)
+    , mCurrentDelay(-1.0)
+    // Round the maximum delay up to the next tick.
+    , mMaxDelayTicks(ceil(aMaxDelayTicks))
+    , mCurrentChunk(0)
+    // mLastReadChunk is initialized in EnsureBuffer
+#ifdef DEBUG
+    , mHaveWrittenBlock(false)
+#endif
+  {
+    // The 180 second limit in AudioContext::CreateDelay() and the
+    // 1 << MEDIA_TIME_FRAC_BITS limit on sample rate provide a limit on the
+    // maximum delay.
+    MOZ_ASSERT(aMaxDelayTicks <=
+               std::numeric_limits<decltype(mMaxDelayTicks)>::max());
+  }
+
+  // Write a WEBAUDIO_BLOCK_SIZE block for aChannelCount channels.
+  void Write(const AudioBlock& aInputChunk);
+
+  // Read a block with an array of delays, in ticks, for each sample frame.
+  // Each delay should be >= 0 and <= MaxDelayTicks().
+  void Read(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+            AudioBlock* aOutputChunk,
+            ChannelInterpretation aChannelInterpretation);
+  // Read a block with a constant delay, which will be smoothed with the
+  // previous delay.  The delay should be >= 0 and <= MaxDelayTicks().
+  void Read(double aDelayTicks, AudioBlock* aOutputChunk,
+            ChannelInterpretation aChannelInterpretation);
+
+  // Read into one of the channels of aOutputChunk, given an array of
+  // delays in ticks.  This is useful when delays are different on different
+  // channels.  aOutputChunk must have already been allocated with at least as
+  // many channels as were in any of the blocks passed to Write().
+  void ReadChannel(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                   AudioBlock* aOutputChunk, uint32_t aChannel,
+                   ChannelInterpretation aChannelInterpretation);
+
+  // Advance the buffer pointer
+  void NextBlock()
+  {
+    mCurrentChunk = (mCurrentChunk + 1) % mChunks.Length();
+#ifdef DEBUG
+    MOZ_ASSERT(mHaveWrittenBlock);
+    mHaveWrittenBlock = false;
+#endif
+  }
+
+  void Reset() {
+    mChunks.Clear();
+    mCurrentDelay = -1.0;
+  };
+
+  int MaxDelayTicks() const { return mMaxDelayTicks; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const;
+
+private:
+  void ReadChannels(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                    AudioBlock* aOutputChunk,
+                    uint32_t aFirstChannel, uint32_t aNumChannelsToRead,
+                    ChannelInterpretation aChannelInterpretation);
+  bool EnsureBuffer();
+  int PositionForDelay(int aDelay);
+  int ChunkForPosition(int aPosition);
+  int OffsetForPosition(int aPosition);
+  int ChunkForDelay(int aDelay);
+  void UpdateUpmixChannels(int aNewReadChunk, uint32_t channelCount,
+                           ChannelInterpretation aChannelInterpretation);
+
+  // Circular buffer for capturing delayed samples.
+  FallibleTArray<AudioChunk> mChunks;
+  // Cache upmixed channel arrays.
+  AutoTArray<const float*,GUESS_AUDIO_CHANNELS> mUpmixChannels;
+  double mSmoothingRate;
+  // Current delay, in fractional ticks
+  double mCurrentDelay;
+  // Maximum delay, in ticks
+  int mMaxDelayTicks;
+  // The current position in the circular buffer.  The next write will be to
+  // this chunk, and the next read may begin before this chunk.
+  int mCurrentChunk;
+  // The chunk owning the pointers in mUpmixChannels
+  int mLastReadChunk;
+#ifdef DEBUG
+  bool mHaveWrittenBlock;
+#endif
+};
+
+} // namespace mozilla
+
+#endif // DelayBuffer_h_
diff --git a/dom/media/webaudio/DelayNode.cpp b/dom/media/webaudio/DelayNode.cpp
new file mode 100644
index 0000000000..17dc725144
--- /dev/null
+++ b/dom/media/webaudio/DelayNode.cpp
@@ -0,0 +1,234 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "DelayNode.h"
+#include "mozilla/dom/DelayNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "DelayBuffer.h"
+#include "PlayingRefChangeHandler.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(DelayNode, AudioNode,
+                                   mDelay)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(DelayNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(DelayNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(DelayNode, AudioNode)
+
+class DelayNodeEngine final : public AudioNodeEngine
+{
+  typedef PlayingRefChangeHandler PlayingRefChanged;
+public:
+  DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                  double aMaxDelayTicks)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default value in sync with the default value in DelayNode::DelayNode.
+    , mDelay(0.f)
+    // Use a smoothing range of 20ms
+    , mBuffer(std::max(aMaxDelayTicks,
+                       static_cast<double>(WEBAUDIO_BLOCK_SIZE)),
+              WebAudioUtils::ComputeSmoothingRate(0.02,
+                                                  mDestination->SampleRate()))
+    , mMaxDelay(aMaxDelayTicks)
+    , mHaveProducedBeforeInput(false)
+    , mLeftOverData(INT32_MIN)
+  {
+  }
+
+  DelayNodeEngine* AsDelayNodeEngine() override
+  {
+    return this;
+  }
+
+  enum Parameters {
+    DELAY,
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case DELAY:
+      mDelay.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad DelayNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    MOZ_ASSERT(aStream->SampleRate() == mDestination->SampleRate());
+
+    if (!aInput.IsSilentOrSubnormal()) {
+      if (mLeftOverData <= 0) {
+        RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aStream, PlayingRefChanged::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      mLeftOverData = mBuffer.MaxDelayTicks();
+    } else if (mLeftOverData > 0) {
+      mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+    } else {
+      if (mLeftOverData != INT32_MIN) {
+        mLeftOverData = INT32_MIN;
+        aStream->ScheduleCheckForInactive();
+
+        // Delete our buffered data now we no longer need it
+        mBuffer.Reset();
+
+        RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aStream, PlayingRefChanged::RELEASE);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    mBuffer.Write(aInput);
+
+    // Skip output update if mLastChunks has already been set by
+    // ProduceBlockBeforeInput() when in a cycle.
+    if (!mHaveProducedBeforeInput) {
+      UpdateOutputBlock(aStream, aFrom, aOutput, 0.0);
+    }
+    mHaveProducedBeforeInput = false;
+    mBuffer.NextBlock();
+  }
+
+  void UpdateOutputBlock(AudioNodeStream* aStream, GraphTime aFrom,
+                         AudioBlock* aOutput, double minDelay)
+  {
+    double maxDelay = mMaxDelay;
+    double sampleRate = aStream->SampleRate();
+    ChannelInterpretation channelInterpretation =
+      aStream->GetChannelInterpretation();
+    if (mDelay.HasSimpleValue()) {
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block, even if that is greater than maxDelay.
+      double delayFrames = mDelay.GetValue() * sampleRate;
+      double delayFramesClamped =
+        std::max(minDelay, std::min(delayFrames, maxDelay));
+      mBuffer.Read(delayFramesClamped, aOutput, channelInterpretation);
+    } else {
+      // Compute the delay values for the duration of the input AudioChunk
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block.
+      StreamTime tick = mDestination->GraphTimeToStreamTime(aFrom);
+      float values[WEBAUDIO_BLOCK_SIZE];
+      mDelay.GetValuesAtTime(tick, values,WEBAUDIO_BLOCK_SIZE);
+
+      double computedDelay[WEBAUDIO_BLOCK_SIZE];
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        double delayAtTick = values[counter] * sampleRate;
+        double delayAtTickClamped =
+          std::max(minDelay, std::min(delayAtTick, maxDelay));
+        computedDelay[counter] = delayAtTickClamped;
+      }
+      mBuffer.Read(computedDelay, aOutput, channelInterpretation);
+    }
+  }
+
+  void ProduceBlockBeforeInput(AudioNodeStream* aStream,
+                               GraphTime aFrom,
+                               AudioBlock* aOutput) override
+  {
+    if (mLeftOverData <= 0) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else {
+      UpdateOutputBlock(aStream, aFrom, aOutput, WEBAUDIO_BLOCK_SIZE);
+    }
+    mHaveProducedBeforeInput = true;
+  }
+
+  bool IsActive() const override
+  {
+    return mLeftOverData != INT32_MIN;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    // Not owned:
+    // - mDestination - probably not owned
+    // - mDelay - shares ref with AudioNode, don't count
+    amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mDelay;
+  DelayBuffer mBuffer;
+  double mMaxDelay;
+  bool mHaveProducedBeforeInput;
+  // How much data we have in our buffer which needs to be flushed out when our inputs
+  // finish.
+  int32_t mLeftOverData;
+};
+
+DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mDelay(new AudioParam(this, DelayNodeEngine::DELAY, 0.0f, "delayTime"))
+{
+  DelayNodeEngine* engine =
+    new DelayNodeEngine(this, aContext->Destination(),
+                        aContext->SampleRate() * aMaxDelay);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+DelayNode::~DelayNode()
+{
+}
+
+size_t
+DelayNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mDelay->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+DelayNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+DelayNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return DelayNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/DelayNode.h b/dom/media/webaudio/DelayNode.h
new file mode 100644
index 0000000000..dfee970bc3
--- /dev/null
+++ b/dom/media/webaudio/DelayNode.h
@@ -0,0 +1,55 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DelayNode_h_
+#define DelayNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class DelayNode final : public AudioNode
+{
+public:
+  DelayNode(AudioContext* aContext, double aMaxDelay);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(DelayNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* DelayTime() const
+  {
+    return mDelay;
+  }
+
+  const char* NodeType() const override
+  {
+    return "DelayNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~DelayNode();
+
+private:
+  friend class DelayNodeEngine;
+
+private:
+  RefPtr<AudioParam> mDelay;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/DynamicsCompressorNode.cpp b/dom/media/webaudio/DynamicsCompressorNode.cpp
new file mode 100644
index 0000000000..3a3dc98491
--- /dev/null
+++ b/dom/media/webaudio/DynamicsCompressorNode.cpp
@@ -0,0 +1,237 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "DynamicsCompressorNode.h"
+#include "mozilla/dom/DynamicsCompressorNodeBinding.h"
+#include "nsAutoPtr.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "blink/DynamicsCompressor.h"
+
+using WebCore::DynamicsCompressor;
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(DynamicsCompressorNode, AudioNode,
+                                   mThreshold,
+                                   mKnee,
+                                   mRatio,
+                                   mAttack,
+                                   mRelease)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(DynamicsCompressorNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(DynamicsCompressorNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(DynamicsCompressorNode, AudioNode)
+
+class DynamicsCompressorNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit DynamicsCompressorNodeEngine(AudioNode* aNode,
+                                        AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default value in sync with the default value in
+    // DynamicsCompressorNode::DynamicsCompressorNode.
+    , mThreshold(-24.f)
+    , mKnee(30.f)
+    , mRatio(12.f)
+    , mAttack(0.003f)
+    , mRelease(0.25f)
+    , mCompressor(new DynamicsCompressor(mDestination->SampleRate(), 2))
+  {
+  }
+
+  enum Parameters {
+    THRESHOLD,
+    KNEE,
+    RATIO,
+    ATTACK,
+    RELEASE
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case THRESHOLD:
+      mThreshold.InsertEvent<int64_t>(aEvent);
+      break;
+    case KNEE:
+      mKnee.InsertEvent<int64_t>(aEvent);
+      break;
+    case RATIO:
+      mRatio.InsertEvent<int64_t>(aEvent);
+      break;
+    case ATTACK:
+      mAttack.InsertEvent<int64_t>(aEvent);
+      break;
+    case RELEASE:
+      mRelease.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad DynamicsCompresssorNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    if (aInput.IsNull()) {
+      // Just output silence
+      *aOutput = aInput;
+      return;
+    }
+
+    const uint32_t channelCount = aInput.ChannelCount();
+    if (mCompressor->numberOfChannels() != channelCount) {
+      // Create a new compressor object with a new channel count
+      mCompressor = new WebCore::DynamicsCompressor(aStream->SampleRate(),
+                                                    aInput.ChannelCount());
+    }
+
+    StreamTime pos = mDestination->GraphTimeToStreamTime(aFrom);
+    mCompressor->setParameterValue(DynamicsCompressor::ParamThreshold,
+                                   mThreshold.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamKnee,
+                                   mKnee.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamRatio,
+                                   mRatio.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamAttack,
+                                   mAttack.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamRelease,
+                                   mRelease.GetValueAtTime(pos));
+
+    aOutput->AllocateChannels(channelCount);
+    mCompressor->process(&aInput, aOutput, aInput.GetDuration());
+
+    SendReductionParamToMainThread(aStream,
+                                   mCompressor->parameterValue(DynamicsCompressor::ParamReduction));
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mDestination (probably)
+    // - Don't count the AudioParamTimelines, their inner refs are owned by the
+    // AudioNode.
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mCompressor->sizeOfIncludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  void SendReductionParamToMainThread(AudioNodeStream* aStream, float aReduction)
+  {
+    MOZ_ASSERT(!NS_IsMainThread());
+
+    class Command final : public Runnable
+    {
+    public:
+      Command(AudioNodeStream* aStream, float aReduction)
+        : mStream(aStream)
+        , mReduction(aReduction)
+      {
+      }
+
+      NS_IMETHOD Run() override
+      {
+        RefPtr<DynamicsCompressorNode> node =
+          static_cast<DynamicsCompressorNode*>
+            (mStream->Engine()->NodeMainThread());
+        if (node) {
+          node->SetReduction(mReduction);
+        }
+        return NS_OK;
+      }
+
+    private:
+      RefPtr<AudioNodeStream> mStream;
+      float mReduction;
+    };
+
+    NS_DispatchToMainThread(new Command(aStream, aReduction));
+  }
+
+private:
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mThreshold;
+  AudioParamTimeline mKnee;
+  AudioParamTimeline mRatio;
+  AudioParamTimeline mAttack;
+  AudioParamTimeline mRelease;
+  nsAutoPtr<DynamicsCompressor> mCompressor;
+};
+
+DynamicsCompressorNode::DynamicsCompressorNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Explicit,
+              ChannelInterpretation::Speakers)
+  , mThreshold(new AudioParam(this, DynamicsCompressorNodeEngine::THRESHOLD,
+                              -24.f, "threshold"))
+  , mKnee(new AudioParam(this, DynamicsCompressorNodeEngine::KNEE,
+                         30.f, "knee"))
+  , mRatio(new AudioParam(this, DynamicsCompressorNodeEngine::RATIO,
+                          12.f, "ratio"))
+  , mReduction(0)
+  , mAttack(new AudioParam(this, DynamicsCompressorNodeEngine::ATTACK,
+                           0.003f, "attack"))
+  , mRelease(new AudioParam(this, DynamicsCompressorNodeEngine::RELEASE,
+                            0.25f, "release"))
+{
+  DynamicsCompressorNodeEngine* engine = new DynamicsCompressorNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+DynamicsCompressorNode::~DynamicsCompressorNode()
+{
+}
+
+size_t
+DynamicsCompressorNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mThreshold->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mKnee->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mRatio->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mAttack->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mRelease->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+DynamicsCompressorNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+DynamicsCompressorNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return DynamicsCompressorNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/DynamicsCompressorNode.h b/dom/media/webaudio/DynamicsCompressorNode.h
new file mode 100644
index 0000000000..5bdd5f2d04
--- /dev/null
+++ b/dom/media/webaudio/DynamicsCompressorNode.h
@@ -0,0 +1,89 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DynamicsCompressorNode_h_
+#define DynamicsCompressorNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class DynamicsCompressorNode final : public AudioNode
+{
+public:
+  explicit DynamicsCompressorNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(DynamicsCompressorNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* Threshold() const
+  {
+    return mThreshold;
+  }
+
+  AudioParam* Knee() const
+  {
+    return mKnee;
+  }
+
+  AudioParam* Ratio() const
+  {
+    return mRatio;
+  }
+
+  AudioParam* Attack() const
+  {
+    return mAttack;
+  }
+
+  // Called GetRelease to prevent clashing with the nsISupports::Release name
+  AudioParam* GetRelease() const
+  {
+    return mRelease;
+  }
+
+  float Reduction() const
+  {
+    return mReduction;
+  }
+
+  const char* NodeType() const override
+  {
+    return "DynamicsCompressorNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void SetReduction(float aReduction)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    mReduction = aReduction;
+  }
+
+protected:
+  virtual ~DynamicsCompressorNode();
+
+private:
+  RefPtr<AudioParam> mThreshold;
+  RefPtr<AudioParam> mKnee;
+  RefPtr<AudioParam> mRatio;
+  float mReduction;
+  RefPtr<AudioParam> mAttack;
+  RefPtr<AudioParam> mRelease;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/FFTBlock.cpp b/dom/media/webaudio/FFTBlock.cpp
new file mode 100644
index 0000000000..f517ef2833
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.cpp
@@ -0,0 +1,226 @@
+/* -*- Mode: C++; tab-width: 4; indent-tabs-mode: nil; c-basic-offset: 4 -*- */
+/* vim:set ts=4 sw=4 sts=4 et cindent: */
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "FFTBlock.h"
+
+#include <complex>
+
+namespace mozilla {
+
+typedef std::complex<double> Complex;
+
+FFTBlock* FFTBlock::CreateInterpolatedBlock(const FFTBlock& block0, const FFTBlock& block1, double interp)
+{
+    FFTBlock* newBlock = new FFTBlock(block0.FFTSize());
+
+    newBlock->InterpolateFrequencyComponents(block0, block1, interp);
+
+    // In the time-domain, the 2nd half of the response must be zero, to avoid circular convolution aliasing...
+    int fftSize = newBlock->FFTSize();
+    AlignedTArray<float> buffer(fftSize);
+    newBlock->GetInverseWithoutScaling(buffer.Elements());
+    AudioBufferInPlaceScale(buffer.Elements(), 1.0f / fftSize, fftSize / 2);
+    PodZero(buffer.Elements() + fftSize / 2, fftSize / 2);
+
+    // Put back into frequency domain.
+    newBlock->PerformFFT(buffer.Elements());
+
+    return newBlock;
+}
+
+void FFTBlock::InterpolateFrequencyComponents(const FFTBlock& block0, const FFTBlock& block1, double interp)
+{
+    // FIXME : with some work, this method could be optimized
+
+    ComplexU* dft = mOutputBuffer.Elements();
+
+    const ComplexU* dft1 = block0.mOutputBuffer.Elements();
+    const ComplexU* dft2 = block1.mOutputBuffer.Elements();
+
+    MOZ_ASSERT(mFFTSize == block0.FFTSize());
+    MOZ_ASSERT(mFFTSize == block1.FFTSize());
+    double s1base = (1.0 - interp);
+    double s2base = interp;
+
+    double phaseAccum = 0.0;
+    double lastPhase1 = 0.0;
+    double lastPhase2 = 0.0;
+
+    int n = mFFTSize / 2;
+
+    dft[0].r = static_cast<float>(s1base * dft1[0].r + s2base * dft2[0].r);
+    dft[n].r = static_cast<float>(s1base * dft1[n].r + s2base * dft2[n].r);
+
+    for (int i = 1; i < n; ++i) {
+        Complex c1(dft1[i].r, dft1[i].i);
+        Complex c2(dft2[i].r, dft2[i].i);
+
+        double mag1 = abs(c1);
+        double mag2 = abs(c2);
+
+        // Interpolate magnitudes in decibels
+        double mag1db = 20.0 * log10(mag1);
+        double mag2db = 20.0 * log10(mag2);
+
+        double s1 = s1base;
+        double s2 = s2base;
+
+        double magdbdiff = mag1db - mag2db;
+
+        // Empirical tweak to retain higher-frequency zeroes
+        double threshold =  (i > 16) ? 5.0 : 2.0;
+
+        if (magdbdiff < -threshold && mag1db < 0.0) {
+            s1 = pow(s1, 0.75);
+            s2 = 1.0 - s1;
+        } else if (magdbdiff > threshold && mag2db < 0.0) {
+            s2 = pow(s2, 0.75);
+            s1 = 1.0 - s2;
+        }
+
+        // Average magnitude by decibels instead of linearly
+        double magdb = s1 * mag1db + s2 * mag2db;
+        double mag = pow(10.0, 0.05 * magdb);
+
+        // Now, deal with phase
+        double phase1 = arg(c1);
+        double phase2 = arg(c2);
+
+        double deltaPhase1 = phase1 - lastPhase1;
+        double deltaPhase2 = phase2 - lastPhase2;
+        lastPhase1 = phase1;
+        lastPhase2 = phase2;
+
+        // Unwrap phase deltas
+        if (deltaPhase1 > M_PI)
+            deltaPhase1 -= 2.0 * M_PI;
+        if (deltaPhase1 < -M_PI)
+            deltaPhase1 += 2.0 * M_PI;
+        if (deltaPhase2 > M_PI)
+            deltaPhase2 -= 2.0 * M_PI;
+        if (deltaPhase2 < -M_PI)
+            deltaPhase2 += 2.0 * M_PI;
+
+        // Blend group-delays
+        double deltaPhaseBlend;
+
+        if (deltaPhase1 - deltaPhase2 > M_PI)
+            deltaPhaseBlend = s1 * deltaPhase1 + s2 * (2.0 * M_PI + deltaPhase2);
+        else if (deltaPhase2 - deltaPhase1 > M_PI)
+            deltaPhaseBlend = s1 * (2.0 * M_PI + deltaPhase1) + s2 * deltaPhase2;
+        else
+            deltaPhaseBlend = s1 * deltaPhase1 + s2 * deltaPhase2;
+
+        phaseAccum += deltaPhaseBlend;
+
+        // Unwrap
+        if (phaseAccum > M_PI)
+            phaseAccum -= 2.0 * M_PI;
+        if (phaseAccum < -M_PI)
+            phaseAccum += 2.0 * M_PI;
+
+        dft[i].r = static_cast<float>(mag * cos(phaseAccum));
+        dft[i].i = static_cast<float>(mag * sin(phaseAccum));
+    }
+}
+
+double FFTBlock::ExtractAverageGroupDelay()
+{
+    ComplexU* dft = mOutputBuffer.Elements();
+
+    double aveSum = 0.0;
+    double weightSum = 0.0;
+    double lastPhase = 0.0;
+
+    int halfSize = FFTSize() / 2;
+
+    const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
+
+    // Remove DC offset
+    dft[0].r = 0.0f;
+
+    // Calculate weighted average group delay
+    for (int i = 1; i < halfSize; i++) {
+        Complex c(dft[i].r, dft[i].i);
+        double mag = abs(c);
+        double phase = arg(c);
+
+        double deltaPhase = phase - lastPhase;
+        lastPhase = phase;
+
+        // Unwrap
+        if (deltaPhase < -M_PI)
+            deltaPhase += 2.0 * M_PI;
+        if (deltaPhase > M_PI)
+            deltaPhase -= 2.0 * M_PI;
+
+        aveSum += mag * deltaPhase;
+        weightSum += mag;
+    }
+
+    // Note how we invert the phase delta wrt frequency since this is how group delay is defined
+    double ave = aveSum / weightSum;
+    double aveSampleDelay = -ave / kSamplePhaseDelay;
+
+    // Leave 20 sample headroom (for leading edge of impulse)
+    aveSampleDelay -= 20.0;
+    if (aveSampleDelay <= 0.0)
+        return 0.0;
+
+    // Remove average group delay (minus 20 samples for headroom)
+    AddConstantGroupDelay(-aveSampleDelay);
+
+    return aveSampleDelay;
+}
+
+void FFTBlock::AddConstantGroupDelay(double sampleFrameDelay)
+{
+    int halfSize = FFTSize() / 2;
+
+    ComplexU* dft = mOutputBuffer.Elements();
+
+    const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
+
+    double phaseAdj = -sampleFrameDelay * kSamplePhaseDelay;
+
+    // Add constant group delay
+    for (int i = 1; i < halfSize; i++) {
+        Complex c(dft[i].r, dft[i].i);
+        double mag = abs(c);
+        double phase = arg(c);
+
+        phase += i * phaseAdj;
+
+        dft[i].r = static_cast<float>(mag * cos(phase));
+        dft[i].i = static_cast<float>(mag * sin(phase));
+    }
+}
+
+} // namespace mozilla
diff --git a/dom/media/webaudio/FFTBlock.h b/dom/media/webaudio/FFTBlock.h
new file mode 100644
index 0000000000..84b9f38aa9
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.h
@@ -0,0 +1,319 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef FFTBlock_h_
+#define FFTBlock_h_
+
+#ifdef BUILD_ARM_NEON
+#include <cmath>
+#include "mozilla/arm.h"
+#include "dl/sp/api/omxSP.h"
+#endif
+
+#include "AlignedTArray.h"
+#include "AudioNodeEngine.h"
+#if defined(MOZ_LIBAV_FFT)
+#ifdef __cplusplus
+extern "C" {
+#endif
+#include "libavcodec/avfft.h"
+#ifdef __cplusplus
+}
+#endif
+#else
+#include "kiss_fft/kiss_fftr.h"
+#endif
+
+namespace mozilla {
+
+// This class defines an FFT block, loosely modeled after Blink's FFTFrame
+// class to make sharing code with Blink easy.
+// Currently it's implemented on top of KissFFT on all platforms.
+class FFTBlock final
+{
+  union ComplexU {
+#if !defined(MOZ_LIBAV_FFT)
+    kiss_fft_cpx c;
+#endif
+    float f[2];
+    struct {
+      float r;
+      float i;
+    };
+  };
+
+public:
+  explicit FFTBlock(uint32_t aFFTSize)
+#if defined(MOZ_LIBAV_FFT)
+    : mAvRDFT(nullptr)
+    , mAvIRDFT(nullptr)
+#else
+    : mKissFFT(nullptr)
+    , mKissIFFT(nullptr)
+#ifdef BUILD_ARM_NEON
+    , mOmxFFT(nullptr)
+    , mOmxIFFT(nullptr)
+#endif
+#endif
+  {
+    MOZ_COUNT_CTOR(FFTBlock);
+    SetFFTSize(aFFTSize);
+  }
+  ~FFTBlock()
+  {
+    MOZ_COUNT_DTOR(FFTBlock);
+    Clear();
+  }
+
+  // Return a new FFTBlock with frequency components interpolated between
+  // |block0| and |block1| with |interp| between 0.0 and 1.0.
+  static FFTBlock*
+  CreateInterpolatedBlock(const FFTBlock& block0,
+                          const FFTBlock& block1, double interp);
+
+  // Transform FFTSize() points of aData and store the result internally.
+  void PerformFFT(const float* aData)
+  {
+    EnsureFFT();
+#if defined(MOZ_LIBAV_FFT)
+    PodCopy(mOutputBuffer.Elements()->f, aData, mFFTSize);
+    av_rdft_calc(mAvRDFT, mOutputBuffer.Elements()->f);
+    // Recover packed Nyquist.
+    mOutputBuffer[mFFTSize / 2].r = mOutputBuffer[0].i;
+    mOutputBuffer[0].i = 0.0f;
+#else
+#ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      omxSP_FFTFwd_RToCCS_F32_Sfs(aData, mOutputBuffer.Elements()->f, mOmxFFT);
+    } else
+#endif
+    {
+      kiss_fftr(mKissFFT, aData, &(mOutputBuffer.Elements()->c));
+    }
+#endif
+  }
+  // Inverse-transform internal data and store the resulting FFTSize()
+  // points in aDataOut.
+  void GetInverse(float* aDataOut)
+  {
+    GetInverseWithoutScaling(aDataOut);
+    AudioBufferInPlaceScale(aDataOut, 1.0f / mFFTSize, mFFTSize);
+  }
+  // Inverse-transform internal frequency data and store the resulting
+  // FFTSize() points in |aDataOut|.  If frequency data has not already been
+  // scaled, then the output will need scaling by 1/FFTSize().
+  void GetInverseWithoutScaling(float* aDataOut)
+  {
+    EnsureIFFT();
+#if defined(MOZ_LIBAV_FFT)
+    {
+      // Even though this function doesn't scale, the libav forward transform
+      // gives a value that needs scaling by 2 in order for things to turn out
+      // similar to how we expect from kissfft/openmax.
+      AudioBufferCopyWithScale(mOutputBuffer.Elements()->f, 2.0f,
+                               aDataOut, mFFTSize);
+      aDataOut[1] = 2.0f * mOutputBuffer[mFFTSize/2].r; // Packed Nyquist
+      av_rdft_calc(mAvIRDFT, aDataOut);
+    }
+#else
+#ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      omxSP_FFTInv_CCSToR_F32_Sfs_unscaled(mOutputBuffer.Elements()->f, aDataOut, mOmxIFFT);
+    } else
+#endif
+    {
+      kiss_fftri(mKissIFFT, &(mOutputBuffer.Elements()->c), aDataOut);
+    }
+#endif
+  }
+
+  void Multiply(const FFTBlock& aFrame)
+  {
+    uint32_t halfSize = mFFTSize / 2;
+    // DFTs are not packed.
+    MOZ_ASSERT(mOutputBuffer[0].i == 0);
+    MOZ_ASSERT(aFrame.mOutputBuffer[0].i == 0);
+
+    BufferComplexMultiply(mOutputBuffer.Elements()->f,
+                          aFrame.mOutputBuffer.Elements()->f,
+                          mOutputBuffer.Elements()->f,
+                          halfSize);
+    mOutputBuffer[halfSize].r *= aFrame.mOutputBuffer[halfSize].r;
+    // This would have been set to NaN if either real component was NaN.
+    mOutputBuffer[0].i = 0.0f;
+  }
+
+  // Perform a forward FFT on |aData|, assuming zeros after dataSize samples,
+  // and pre-scale the generated internal frequency domain coefficients so
+  // that GetInverseWithoutScaling() can be used to transform to the time
+  // domain.  This is useful for convolution kernels.
+  void PadAndMakeScaledDFT(const float* aData, size_t dataSize)
+  {
+    MOZ_ASSERT(dataSize <= FFTSize());
+    AlignedTArray<float> paddedData;
+    paddedData.SetLength(FFTSize());
+    AudioBufferCopyWithScale(aData, 1.0f / FFTSize(),
+                             paddedData.Elements(), dataSize);
+    PodZero(paddedData.Elements() + dataSize, mFFTSize - dataSize);
+    PerformFFT(paddedData.Elements());
+  }
+
+  void SetFFTSize(uint32_t aSize)
+  {
+    mFFTSize = aSize;
+    mOutputBuffer.SetLength(aSize / 2 + 1);
+    PodZero(mOutputBuffer.Elements(), aSize / 2 + 1);
+    Clear();
+  }
+
+  // Return the average group delay and removes this from the frequency data.
+  double ExtractAverageGroupDelay();
+
+  uint32_t FFTSize() const
+  {
+    return mFFTSize;
+  }
+  float RealData(uint32_t aIndex) const
+  {
+    return mOutputBuffer[aIndex].r;
+  }
+  float& RealData(uint32_t aIndex)
+  {
+    return mOutputBuffer[aIndex].r;
+  }
+  float ImagData(uint32_t aIndex) const
+  {
+    return mOutputBuffer[aIndex].i;
+  }
+  float& ImagData(uint32_t aIndex)
+  {
+    return mOutputBuffer[aIndex].i;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    size_t amount = 0;
+#if defined(MOZ_LIBAV_FFT)
+    amount += aMallocSizeOf(mAvRDFT);
+    amount += aMallocSizeOf(mAvIRDFT);
+#else
+    amount += aMallocSizeOf(mKissFFT);
+    amount += aMallocSizeOf(mKissIFFT);
+#endif
+    amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  FFTBlock(const FFTBlock& other) = delete;
+  void operator=(const FFTBlock& other) = delete;
+
+  void EnsureFFT()
+  {
+#if defined(MOZ_LIBAV_FFT)
+    if (!mAvRDFT) {
+      mAvRDFT = av_rdft_init(log((double)mFFTSize)/M_LN2, DFT_R2C);
+    }
+#else
+#ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      if (!mOmxFFT) {
+        mOmxFFT = createOmxFFT(mFFTSize);
+      }
+    } else
+#endif
+    {
+      if (!mKissFFT) {
+        mKissFFT = kiss_fftr_alloc(mFFTSize, 0, nullptr, nullptr);
+      }
+    }
+#endif
+  }
+  void EnsureIFFT()
+  {
+#if defined(MOZ_LIBAV_FFT)
+    if (!mAvIRDFT) {
+      mAvIRDFT = av_rdft_init(log((double)mFFTSize)/M_LN2, IDFT_C2R);
+    }
+#else
+#ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      if (!mOmxIFFT) {
+        mOmxIFFT = createOmxFFT(mFFTSize);
+      }
+    } else
+#endif
+    {
+      if (!mKissIFFT) {
+        mKissIFFT = kiss_fftr_alloc(mFFTSize, 1, nullptr, nullptr);
+      }
+    }
+#endif
+  }
+
+#ifdef BUILD_ARM_NEON
+  static OMXFFTSpec_R_F32* createOmxFFT(uint32_t aFFTSize)
+  {
+    MOZ_ASSERT((aFFTSize & (aFFTSize-1)) == 0);
+    OMX_INT bufSize;
+    OMX_INT order = log((double)aFFTSize)/M_LN2;
+    MOZ_ASSERT(aFFTSize>>order == 1);
+    OMXResult status = omxSP_FFTGetBufSize_R_F32(order, &bufSize);
+    if (status == OMX_Sts_NoErr) {
+      OMXFFTSpec_R_F32* context = static_cast<OMXFFTSpec_R_F32*>(malloc(bufSize));
+      if (omxSP_FFTInit_R_F32(context, order) != OMX_Sts_NoErr) {
+        return nullptr;
+      }
+      return context;
+    }
+    return nullptr;
+  }
+#endif
+
+  void Clear()
+  {
+#if defined(MOZ_LIBAV_FFT)
+    av_rdft_end(mAvRDFT);
+    av_rdft_end(mAvIRDFT);
+    mAvRDFT = mAvIRDFT = nullptr;
+#else
+#ifdef BUILD_ARM_NEON
+    free(mOmxFFT);
+    free(mOmxIFFT);
+    mOmxFFT = mOmxIFFT = nullptr;
+#endif
+    free(mKissFFT);
+    free(mKissIFFT);
+    mKissFFT = mKissIFFT = nullptr;
+#endif
+  }
+  void AddConstantGroupDelay(double sampleFrameDelay);
+  void InterpolateFrequencyComponents(const FFTBlock& block0,
+                                      const FFTBlock& block1, double interp);
+#if defined(MOZ_LIBAV_FFT)
+  RDFTContext *mAvRDFT;
+  RDFTContext *mAvIRDFT;
+#else
+  kiss_fftr_cfg mKissFFT;
+  kiss_fftr_cfg mKissIFFT;
+#ifdef BUILD_ARM_NEON
+  OMXFFTSpec_R_F32* mOmxFFT;
+  OMXFFTSpec_R_F32* mOmxIFFT;
+#endif
+#endif
+  AlignedTArray<ComplexU> mOutputBuffer;
+  uint32_t mFFTSize;
+};
+
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/GainNode.cpp b/dom/media/webaudio/GainNode.cpp
new file mode 100644
index 0000000000..46ac997638
--- /dev/null
+++ b/dom/media/webaudio/GainNode.cpp
@@ -0,0 +1,156 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "GainNode.h"
+#include "mozilla/dom/GainNodeBinding.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(GainNode, AudioNode,
+                                   mGain)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(GainNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(GainNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(GainNode, AudioNode)
+
+class GainNodeEngine final : public AudioNodeEngine
+{
+public:
+  GainNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default value in sync with the default value in GainNode::GainNode.
+    , mGain(1.f)
+  {
+  }
+
+  enum Parameters {
+    GAIN
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case GAIN:
+      mGain.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad GainNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    if (aInput.IsNull()) {
+      // If input is silent, so is the output
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else if (mGain.HasSimpleValue()) {
+      // Optimize the case where we only have a single value set as the volume
+      float gain = mGain.GetValue();
+      if (gain == 0.0f) {
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      } else {
+        *aOutput = aInput;
+        aOutput->mVolume *= gain;
+      }
+    } else {
+      // First, compute a vector of gains for each track tick based on the
+      // timeline at hand, and then for each channel, multiply the values
+      // in the buffer with the gain vector.
+      aOutput->AllocateChannels(aInput.ChannelCount());
+
+      // Compute the gain values for the duration of the input AudioChunk
+      StreamTime tick = mDestination->GraphTimeToStreamTime(aFrom);
+      float computedGain[WEBAUDIO_BLOCK_SIZE + 4];
+      float* alignedComputedGain = ALIGNED16(computedGain);
+      ASSERT_ALIGNED16(alignedComputedGain);
+      mGain.GetValuesAtTime(tick, alignedComputedGain, WEBAUDIO_BLOCK_SIZE);
+
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        alignedComputedGain[counter] *= aInput.mVolume;
+      }
+
+      // Apply the gain to the output buffer
+      for (size_t channel = 0; channel < aOutput->ChannelCount(); ++channel) {
+        const float* inputBuffer = static_cast<const float*> (aInput.mChannelData[channel]);
+        float* buffer = aOutput->ChannelFloatsForWrite(channel);
+        AudioBlockCopyChannelWithScale(inputBuffer, alignedComputedGain, buffer);
+      }
+    }
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mDestination (probably)
+    // - mGain - Internal ref owned by AudioNode
+    return AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mGain;
+};
+
+GainNode::GainNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mGain(new AudioParam(this, GainNodeEngine::GAIN, 1.0f, "gain"))
+{
+  GainNodeEngine* engine = new GainNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+GainNode::~GainNode()
+{
+}
+
+size_t
+GainNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mGain->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+GainNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+GainNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return GainNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/GainNode.h b/dom/media/webaudio/GainNode.h
new file mode 100644
index 0000000000..aab22ad65e
--- /dev/null
+++ b/dom/media/webaudio/GainNode.h
@@ -0,0 +1,52 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef GainNode_h_
+#define GainNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class GainNode final : public AudioNode
+{
+public:
+  explicit GainNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(GainNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* Gain() const
+  {
+    return mGain;
+  }
+
+  const char* NodeType() const override
+  {
+    return "GainNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~GainNode();
+
+private:
+  RefPtr<AudioParam> mGain;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/IIRFilterNode.cpp b/dom/media/webaudio/IIRFilterNode.cpp
new file mode 100644
index 0000000000..3a69a94c8c
--- /dev/null
+++ b/dom/media/webaudio/IIRFilterNode.cpp
@@ -0,0 +1,228 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "IIRFilterNode.h"
+#include "AudioNodeEngine.h"
+
+#include "blink/IIRFilter.h"
+
+#include "nsGkAtoms.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_ISUPPORTS_INHERITED0(IIRFilterNode, AudioNode)
+
+class IIRFilterNodeEngine final : public AudioNodeEngine
+{
+public:
+  IIRFilterNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                      const AudioDoubleArray &aFeedforward,
+                      const AudioDoubleArray &aFeedback,
+                      uint64_t aWindowID)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    , mFeedforward(aFeedforward)
+    , mFeedback(aFeedback)
+    , mWindowID(aWindowID)
+  {
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    float inputBuffer[WEBAUDIO_BLOCK_SIZE + 4];
+    float* alignedInputBuffer = ALIGNED16(inputBuffer);
+    ASSERT_ALIGNED16(alignedInputBuffer);
+
+    if (aInput.IsNull()) {
+      if (!mIIRFilters.IsEmpty()) {
+        bool allZero = true;
+        for (uint32_t i = 0; i < mIIRFilters.Length(); ++i) {
+          allZero &= mIIRFilters[i]->buffersAreZero();
+        }
+
+        // all filter buffer values are zero, so the output will be zero
+        // as well.
+        if (allZero) {
+          mIIRFilters.Clear();
+          aStream->ScheduleCheckForInactive();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::RELEASE);
+          aStream->Graph()->
+            DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+
+          aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+          return;
+        }
+
+        PodZero(alignedInputBuffer, WEBAUDIO_BLOCK_SIZE);
+      }
+    } else if(mIIRFilters.Length() != aInput.ChannelCount()){
+      if (mIIRFilters.IsEmpty()) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+          new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      } else {
+        WebAudioUtils::LogToDeveloperConsole(mWindowID,
+                                             "IIRFilterChannelCountChangeWarning");
+      }
+
+      // Adjust the number of filters based on the number of channels
+      mIIRFilters.SetLength(aInput.ChannelCount());
+      for (size_t i = 0; i < aInput.ChannelCount(); ++i) {
+        mIIRFilters[i] = new blink::IIRFilter(&mFeedforward, &mFeedback);
+      }
+    }
+
+    uint32_t numberOfChannels = mIIRFilters.Length();
+    aOutput->AllocateChannels(numberOfChannels);
+
+    for (uint32_t i = 0; i < numberOfChannels; ++i) {
+      const float* input;
+      if (aInput.IsNull()) {
+        input = alignedInputBuffer;
+      } else {
+        input = static_cast<const float*>(aInput.mChannelData[i]);
+        if (aInput.mVolume != 1.0) {
+          AudioBlockCopyChannelWithScale(input, aInput.mVolume, alignedInputBuffer);
+          input = alignedInputBuffer;
+        }
+      }
+
+      mIIRFilters[i]->process(input,
+                              aOutput->ChannelFloatsForWrite(i),
+                              aInput.GetDuration());
+    }
+  }
+
+  bool IsActive() const override
+  {
+    return !mIIRFilters.IsEmpty();
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mDestination - probably not owned
+    // - AudioParamTimelines - counted in the AudioNode
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mIIRFilters.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  AudioNodeStream* mDestination;
+  nsTArray<nsAutoPtr<blink::IIRFilter>> mIIRFilters;
+  AudioDoubleArray mFeedforward;
+  AudioDoubleArray mFeedback;
+  uint64_t mWindowID;
+};
+
+IIRFilterNode::IIRFilterNode(AudioContext* aContext,
+                             const mozilla::dom::binding_detail::AutoSequence<double>& aFeedforward,
+                             const mozilla::dom::binding_detail::AutoSequence<double>& aFeedback)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+{
+  mFeedforward.SetLength(aFeedforward.Length());
+  PodCopy(mFeedforward.Elements(), aFeedforward.Elements(), aFeedforward.Length());
+  mFeedback.SetLength(aFeedback.Length());
+  PodCopy(mFeedback.Elements(), aFeedback.Elements(), aFeedback.Length());
+
+  // Scale coefficients -- we guarantee that mFeedback != 0 when creating
+  // the IIRFilterNode.
+  double scale = mFeedback[0];
+  double* elements = mFeedforward.Elements();
+  for (size_t i = 0; i < mFeedforward.Length(); ++i) {
+    elements[i] /= scale;
+  }
+
+  elements = mFeedback.Elements();
+  for (size_t i = 0; i < mFeedback.Length(); ++i) {
+    elements[i] /= scale;
+  }
+
+  // We check that this is exactly equal to one later in blink/IIRFilter.cpp
+  elements[0] = 1.0;
+
+  uint64_t windowID = aContext->GetParentObject()->WindowID();
+  IIRFilterNodeEngine* engine = new IIRFilterNodeEngine(this, aContext->Destination(), mFeedforward, mFeedback, windowID);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+IIRFilterNode::~IIRFilterNode()
+{
+}
+
+size_t
+IIRFilterNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+IIRFilterNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+IIRFilterNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return IIRFilterNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+IIRFilterNode::GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                                    const Float32Array& aMagResponse,
+                                    const Float32Array& aPhaseResponse)
+{
+  aFrequencyHz.ComputeLengthAndData();
+  aMagResponse.ComputeLengthAndData();
+  aPhaseResponse.ComputeLengthAndData();
+
+  uint32_t length = std::min(std::min(aFrequencyHz.Length(),
+                                      aMagResponse.Length()),
+                             aPhaseResponse.Length());
+  if (!length) {
+    return;
+  }
+
+  auto frequencies = MakeUnique<float[]>(length);
+  float* frequencyHz = aFrequencyHz.Data();
+  const double nyquist = Context()->SampleRate() * 0.5;
+
+  // Normalize the frequencies
+  for (uint32_t i = 0; i < length; ++i) {
+    if (frequencyHz[i] >= 0 && frequencyHz[i] <= nyquist) {
+        frequencies[i] = static_cast<float>(frequencyHz[i] / nyquist);
+    } else {
+        frequencies[i] = std::numeric_limits<float>::quiet_NaN();
+    }
+  }
+
+  blink::IIRFilter filter(&mFeedforward, &mFeedback);
+  filter.getFrequencyResponse(int(length), frequencies.get(), aMagResponse.Data(), aPhaseResponse.Data());
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/IIRFilterNode.h b/dom/media/webaudio/IIRFilterNode.h
new file mode 100644
index 0000000000..78546c3e54
--- /dev/null
+++ b/dom/media/webaudio/IIRFilterNode.h
@@ -0,0 +1,55 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef IIRFilterNode_h_
+#define IIRFilterNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/IIRFilterNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class IIRFilterNode final : public AudioNode
+{
+public:
+  explicit IIRFilterNode(AudioContext* aContext,
+                         const mozilla::dom::binding_detail::AutoSequence<double>& aFeedforward,
+                         const mozilla::dom::binding_detail::AutoSequence<double>& aFeedback);
+
+  NS_DECL_ISUPPORTS_INHERITED
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+
+  void GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                            const Float32Array& aMagResponse,
+                            const Float32Array& aPhaseResponse);
+
+  const char* NodeType() const override
+  {
+    return "IIRFilterNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~IIRFilterNode();
+
+private:
+    nsTArray<double> mFeedback;
+    nsTArray<double> mFeedforward;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/MediaBufferDecoder.cpp b/dom/media/webaudio/MediaBufferDecoder.cpp
new file mode 100644
index 0000000000..e9f1d5a47c
--- /dev/null
+++ b/dom/media/webaudio/MediaBufferDecoder.cpp
@@ -0,0 +1,649 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaBufferDecoder.h"
+#include "BufferDecoder.h"
+#include "mozilla/dom/AudioContextBinding.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include <speex/speex_resampler.h>
+#include "nsXPCOMCIDInternal.h"
+#include "nsComponentManagerUtils.h"
+#include "MediaDecoderReader.h"
+#include "BufferMediaResource.h"
+#include "DecoderTraits.h"
+#include "AudioContext.h"
+#include "AudioBuffer.h"
+#include "nsContentUtils.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsIScriptError.h"
+#include "nsMimeTypes.h"
+#include "VideoUtils.h"
+#include "WebAudioUtils.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/Telemetry.h"
+#include "nsPrintfCString.h"
+#include "GMPService.h"
+
+namespace mozilla {
+
+extern LazyLogModule gMediaDecoderLog;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(WebAudioDecodeJob)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(WebAudioDecodeJob)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mContext)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mOutput)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mSuccessCallback)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mFailureCallback)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(WebAudioDecodeJob)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mContext)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOutput)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mSuccessCallback)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mFailureCallback)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(WebAudioDecodeJob)
+NS_IMPL_CYCLE_COLLECTION_TRACE_END
+NS_IMPL_CYCLE_COLLECTION_ROOT_NATIVE(WebAudioDecodeJob, AddRef)
+NS_IMPL_CYCLE_COLLECTION_UNROOT_NATIVE(WebAudioDecodeJob, Release)
+
+using namespace dom;
+
+class ReportResultTask final : public Runnable
+{
+public:
+  ReportResultTask(WebAudioDecodeJob& aDecodeJob,
+                   WebAudioDecodeJob::ResultFn aFunction,
+                   WebAudioDecodeJob::ErrorCode aErrorCode)
+    : mDecodeJob(aDecodeJob)
+    , mFunction(aFunction)
+    , mErrorCode(aErrorCode)
+  {
+    MOZ_ASSERT(aFunction);
+  }
+
+  NS_IMETHOD Run() override
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    (mDecodeJob.*mFunction)(mErrorCode);
+
+    return NS_OK;
+  }
+
+private:
+  // Note that the mDecodeJob member will probably die when mFunction is run.
+  // Therefore, it is not safe to do anything fancy with it in this class.
+  // Really, this class is only used because nsRunnableMethod doesn't support
+  // methods accepting arguments.
+  WebAudioDecodeJob& mDecodeJob;
+  WebAudioDecodeJob::ResultFn mFunction;
+  WebAudioDecodeJob::ErrorCode mErrorCode;
+};
+
+enum class PhaseEnum : int
+{
+  Decode,
+  AllocateBuffer,
+  Done
+};
+
+class MediaDecodeTask final : public Runnable
+{
+public:
+  MediaDecodeTask(const char* aContentType, uint8_t* aBuffer,
+                  uint32_t aLength,
+                  WebAudioDecodeJob& aDecodeJob)
+    : mContentType(aContentType)
+    , mBuffer(aBuffer)
+    , mLength(aLength)
+    , mDecodeJob(aDecodeJob)
+    , mPhase(PhaseEnum::Decode)
+    , mFirstFrameDecoded(false)
+  {
+    MOZ_ASSERT(aBuffer);
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  NS_IMETHOD Run();
+  bool CreateReader();
+  MediaDecoderReader* Reader() { MOZ_ASSERT(mDecoderReader); return mDecoderReader; }
+
+private:
+  void ReportFailureOnMainThread(WebAudioDecodeJob::ErrorCode aErrorCode) {
+    if (NS_IsMainThread()) {
+      Cleanup();
+      mDecodeJob.OnFailure(aErrorCode);
+    } else {
+      // Take extra care to cleanup on the main thread
+      NS_DispatchToMainThread(NewRunnableMethod(this, &MediaDecodeTask::Cleanup));
+
+      nsCOMPtr<nsIRunnable> event =
+        new ReportResultTask(mDecodeJob, &WebAudioDecodeJob::OnFailure, aErrorCode);
+      NS_DispatchToMainThread(event);
+    }
+  }
+
+  void Decode();
+  void OnMetadataRead(MetadataHolder* aMetadata);
+  void OnMetadataNotRead(const MediaResult& aError);
+  void RequestSample();
+  void SampleDecoded(MediaData* aData);
+  void SampleNotDecoded(const MediaResult& aError);
+  void FinishDecode();
+  void AllocateBuffer();
+  void CallbackTheResult();
+
+  void Cleanup()
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    // MediaDecoderReader expects that BufferDecoder is alive.
+    // Destruct MediaDecoderReader first.
+    mDecoderReader = nullptr;
+    mBufferDecoder = nullptr;
+    JS_free(nullptr, mBuffer);
+  }
+
+private:
+  nsCString mContentType;
+  uint8_t* mBuffer;
+  uint32_t mLength;
+  WebAudioDecodeJob& mDecodeJob;
+  PhaseEnum mPhase;
+  RefPtr<BufferDecoder> mBufferDecoder;
+  RefPtr<MediaDecoderReader> mDecoderReader;
+  MediaInfo mMediaInfo;
+  MediaQueue<MediaData> mAudioQueue;
+  bool mFirstFrameDecoded;
+};
+
+NS_IMETHODIMP
+MediaDecodeTask::Run()
+{
+  MOZ_ASSERT(mBufferDecoder);
+  MOZ_ASSERT(mDecoderReader);
+  switch (mPhase) {
+  case PhaseEnum::Decode:
+    Decode();
+    break;
+  case PhaseEnum::AllocateBuffer:
+    AllocateBuffer();
+    break;
+  case PhaseEnum::Done:
+    break;
+  }
+
+  return NS_OK;
+}
+
+class BufferDecoderGMPCrashHelper : public GMPCrashHelper
+{
+public:
+  explicit BufferDecoderGMPCrashHelper(nsPIDOMWindowInner* aParent)
+    : mParent(do_GetWeakReference(aParent))
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+  already_AddRefed<nsPIDOMWindowInner> GetPluginCrashedEventTarget() override
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    nsCOMPtr<nsPIDOMWindowInner> window = do_QueryReferent(mParent);
+    return window.forget();
+  }
+private:
+  nsWeakPtr mParent;
+};
+
+bool
+MediaDecodeTask::CreateReader()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+
+  nsCOMPtr<nsIPrincipal> principal;
+  nsCOMPtr<nsIScriptObjectPrincipal> sop = do_QueryInterface(mDecodeJob.mContext->GetParentObject());
+  if (sop) {
+    principal = sop->GetPrincipal();
+  }
+
+  RefPtr<BufferMediaResource> resource =
+    new BufferMediaResource(static_cast<uint8_t*> (mBuffer),
+                            mLength, principal, mContentType);
+
+  MOZ_ASSERT(!mBufferDecoder);
+  mBufferDecoder = new BufferDecoder(resource,
+    new BufferDecoderGMPCrashHelper(mDecodeJob.mContext->GetParentObject()));
+
+  // If you change this list to add support for new decoders, please consider
+  // updating HTMLMediaElement::CreateDecoder as well.
+
+  mDecoderReader = DecoderTraits::CreateReader(mContentType, mBufferDecoder);
+
+  if (!mDecoderReader) {
+    return false;
+  }
+
+  nsresult rv = mDecoderReader->Init();
+  if (NS_FAILED(rv)) {
+    return false;
+  }
+
+  return true;
+}
+
+class AutoResampler final
+{
+public:
+  AutoResampler()
+    : mResampler(nullptr)
+  {}
+  ~AutoResampler()
+  {
+    if (mResampler) {
+      speex_resampler_destroy(mResampler);
+    }
+  }
+  operator SpeexResamplerState*() const
+  {
+    MOZ_ASSERT(mResampler);
+    return mResampler;
+  }
+  void operator=(SpeexResamplerState* aResampler)
+  {
+    mResampler = aResampler;
+  }
+
+private:
+  SpeexResamplerState* mResampler;
+};
+
+void
+MediaDecodeTask::Decode()
+{
+  MOZ_ASSERT(!NS_IsMainThread());
+
+  mBufferDecoder->BeginDecoding(mDecoderReader->OwnerThread());
+
+  // Tell the decoder reader that we are not going to play the data directly,
+  // and that we should not reject files with more channels than the audio
+  // backend support.
+  mDecoderReader->SetIgnoreAudioOutputFormat();
+
+  mDecoderReader->AsyncReadMetadata()->Then(mDecoderReader->OwnerThread(), __func__, this,
+                                       &MediaDecodeTask::OnMetadataRead,
+                                       &MediaDecodeTask::OnMetadataNotRead);
+}
+
+void
+MediaDecodeTask::OnMetadataRead(MetadataHolder* aMetadata)
+{
+  mMediaInfo = aMetadata->mInfo;
+  if (!mMediaInfo.HasAudio()) {
+    mDecoderReader->Shutdown();
+    ReportFailureOnMainThread(WebAudioDecodeJob::NoAudio);
+    return;
+  }
+
+  nsCString codec;
+  if (!mMediaInfo.mAudio.GetAsAudioInfo()->mMimeType.IsEmpty()) {
+    codec = nsPrintfCString("webaudio; %s", mMediaInfo.mAudio.GetAsAudioInfo()->mMimeType.get());
+  } else {
+    codec = nsPrintfCString("webaudio;resource; %s", mContentType.get());
+  }
+
+  nsCOMPtr<nsIRunnable> task = NS_NewRunnableFunction([codec]() -> void {
+    MOZ_ASSERT(!codec.IsEmpty());
+    MOZ_LOG(gMediaDecoderLog,
+            LogLevel::Debug,
+            ("Telemetry (WebAudio) MEDIA_CODEC_USED= '%s'", codec.get()));
+    Telemetry::Accumulate(Telemetry::ID::MEDIA_CODEC_USED, codec);
+  });
+  AbstractThread::MainThread()->Dispatch(task.forget());
+
+  RequestSample();
+}
+
+void
+MediaDecodeTask::OnMetadataNotRead(const MediaResult& aReason)
+{
+  mDecoderReader->Shutdown();
+  ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+}
+
+void
+MediaDecodeTask::RequestSample()
+{
+  mDecoderReader->RequestAudioData()->Then(mDecoderReader->OwnerThread(), __func__, this,
+                                           &MediaDecodeTask::SampleDecoded,
+                                           &MediaDecodeTask::SampleNotDecoded);
+}
+
+void
+MediaDecodeTask::SampleDecoded(MediaData* aData)
+{
+  MOZ_ASSERT(!NS_IsMainThread());
+  mAudioQueue.Push(aData);
+  if (!mFirstFrameDecoded) {
+    mDecoderReader->ReadUpdatedMetadata(&mMediaInfo);
+    mFirstFrameDecoded = true;
+  }
+  RequestSample();
+}
+
+void
+MediaDecodeTask::SampleNotDecoded(const MediaResult& aError)
+{
+  MOZ_ASSERT(!NS_IsMainThread());
+  if (aError == NS_ERROR_DOM_MEDIA_END_OF_STREAM) {
+    FinishDecode();
+  } else {
+    mDecoderReader->Shutdown();
+    ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+  }
+}
+
+void
+MediaDecodeTask::FinishDecode()
+{
+  mDecoderReader->Shutdown();
+
+  uint32_t frameCount = mAudioQueue.FrameCount();
+  uint32_t channelCount = mMediaInfo.mAudio.mChannels;
+  uint32_t sampleRate = mMediaInfo.mAudio.mRate;
+
+  if (!frameCount || !channelCount || !sampleRate) {
+    ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+    return;
+  }
+
+  const uint32_t destSampleRate = mDecodeJob.mContext->SampleRate();
+  AutoResampler resampler;
+
+  uint32_t resampledFrames = frameCount;
+  if (sampleRate != destSampleRate) {
+    resampledFrames = static_cast<uint32_t>(
+        static_cast<uint64_t>(destSampleRate) *
+        static_cast<uint64_t>(frameCount) /
+        static_cast<uint64_t>(sampleRate)
+      );
+
+    resampler = speex_resampler_init(channelCount,
+                                     sampleRate,
+                                     destSampleRate,
+                                     SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
+    speex_resampler_skip_zeros(resampler);
+    resampledFrames += speex_resampler_get_output_latency(resampler);
+  }
+
+  // Allocate the channel buffers.  Note that if we end up resampling, we may
+  // write fewer bytes than mResampledFrames to the output buffer, in which
+  // case mWriteIndex will tell us how many valid samples we have.
+  mDecodeJob.mBuffer = ThreadSharedFloatArrayBufferList::
+    Create(channelCount, resampledFrames, fallible);
+  if (!mDecodeJob.mBuffer) {
+    ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
+    return;
+  }
+
+  RefPtr<MediaData> mediaData;
+  while ((mediaData = mAudioQueue.PopFront())) {
+    RefPtr<AudioData> audioData = mediaData->As<AudioData>();
+    audioData->EnsureAudioBuffer(); // could lead to a copy :(
+    AudioDataValue* bufferData = static_cast<AudioDataValue*>
+      (audioData->mAudioBuffer->Data());
+
+    if (sampleRate != destSampleRate) {
+      const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
+
+      for (uint32_t i = 0; i < audioData->mChannels; ++i) {
+        uint32_t inSamples = audioData->mFrames;
+        uint32_t outSamples = maxOutSamples;
+        float* outData =
+          mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
+
+        WebAudioUtils::SpeexResamplerProcess(
+            resampler, i, &bufferData[i * audioData->mFrames], &inSamples,
+            outData, &outSamples);
+
+        if (i == audioData->mChannels - 1) {
+          mDecodeJob.mWriteIndex += outSamples;
+          MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
+          MOZ_ASSERT(inSamples == audioData->mFrames);
+        }
+      }
+    } else {
+      for (uint32_t i = 0; i < audioData->mChannels; ++i) {
+        float* outData =
+          mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
+        ConvertAudioSamples(&bufferData[i * audioData->mFrames],
+                            outData, audioData->mFrames);
+
+        if (i == audioData->mChannels - 1) {
+          mDecodeJob.mWriteIndex += audioData->mFrames;
+        }
+      }
+    }
+  }
+
+  if (sampleRate != destSampleRate) {
+    uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
+    const uint32_t maxOutSamples = resampledFrames - mDecodeJob.mWriteIndex;
+    for (uint32_t i = 0; i < channelCount; ++i) {
+      uint32_t inSamples = inputLatency;
+      uint32_t outSamples = maxOutSamples;
+      float* outData =
+        mDecodeJob.mBuffer->GetDataForWrite(i) + mDecodeJob.mWriteIndex;
+
+      WebAudioUtils::SpeexResamplerProcess(
+          resampler, i, (AudioDataValue*)nullptr, &inSamples,
+          outData, &outSamples);
+
+      if (i == channelCount - 1) {
+        mDecodeJob.mWriteIndex += outSamples;
+        MOZ_ASSERT(mDecodeJob.mWriteIndex <= resampledFrames);
+        MOZ_ASSERT(inSamples == inputLatency);
+      }
+    }
+  }
+
+  mPhase = PhaseEnum::AllocateBuffer;
+  NS_DispatchToMainThread(this);
+}
+
+void
+MediaDecodeTask::AllocateBuffer()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mDecodeJob.AllocateBuffer()) {
+    ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
+    return;
+  }
+
+  mPhase = PhaseEnum::Done;
+  CallbackTheResult();
+}
+
+void
+MediaDecodeTask::CallbackTheResult()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+  Cleanup();
+
+  // Now, we're ready to call the script back with the resulting buffer
+  mDecodeJob.OnSuccess(WebAudioDecodeJob::NoError);
+}
+
+bool
+WebAudioDecodeJob::AllocateBuffer()
+{
+  MOZ_ASSERT(!mOutput);
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // Now create the AudioBuffer
+  ErrorResult rv;
+  uint32_t channelCount = mBuffer->GetChannels();
+  mOutput = AudioBuffer::Create(mContext, channelCount,
+                                mWriteIndex, mContext->SampleRate(),
+                                mBuffer.forget(), rv);
+  return !rv.Failed();
+}
+
+void
+AsyncDecodeWebAudio(const char* aContentType, uint8_t* aBuffer,
+                    uint32_t aLength, WebAudioDecodeJob& aDecodeJob)
+{
+  // Do not attempt to decode the media if we were not successful at sniffing
+  // the content type.
+  if (!*aContentType ||
+      strcmp(aContentType, APPLICATION_OCTET_STREAM) == 0) {
+    nsCOMPtr<nsIRunnable> event =
+      new ReportResultTask(aDecodeJob,
+                           &WebAudioDecodeJob::OnFailure,
+                           WebAudioDecodeJob::UnknownContent);
+    JS_free(nullptr, aBuffer);
+    NS_DispatchToMainThread(event);
+    return;
+  }
+
+  RefPtr<MediaDecodeTask> task =
+    new MediaDecodeTask(aContentType, aBuffer, aLength, aDecodeJob);
+  if (!task->CreateReader()) {
+    nsCOMPtr<nsIRunnable> event =
+      new ReportResultTask(aDecodeJob,
+                           &WebAudioDecodeJob::OnFailure,
+                           WebAudioDecodeJob::UnknownError);
+    NS_DispatchToMainThread(event);
+  } else {
+    // If we did this without a temporary:
+    //   task->Reader()->OwnerThread()->Dispatch(task.forget())
+    // we might evaluate the task.forget() before calling Reader(). Enforce
+    // a non-crashy order-of-operations.
+    TaskQueue* taskQueue = task->Reader()->OwnerThread();
+    taskQueue->Dispatch(task.forget());
+  }
+}
+
+WebAudioDecodeJob::WebAudioDecodeJob(const nsACString& aContentType,
+                                     AudioContext* aContext,
+                                     Promise* aPromise,
+                                     DecodeSuccessCallback* aSuccessCallback,
+                                     DecodeErrorCallback* aFailureCallback)
+  : mContentType(aContentType)
+  , mWriteIndex(0)
+  , mContext(aContext)
+  , mPromise(aPromise)
+  , mSuccessCallback(aSuccessCallback)
+  , mFailureCallback(aFailureCallback)
+{
+  MOZ_ASSERT(aContext);
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_CTOR(WebAudioDecodeJob);
+}
+
+WebAudioDecodeJob::~WebAudioDecodeJob()
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_DTOR(WebAudioDecodeJob);
+}
+
+void
+WebAudioDecodeJob::OnSuccess(ErrorCode aErrorCode)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aErrorCode == NoError);
+
+  if (mSuccessCallback) {
+    ErrorResult rv;
+    mSuccessCallback->Call(*mOutput, rv);
+    // Ignore errors in calling the callback, since there is not much that we can
+    // do about it here.
+    rv.SuppressException();
+  }
+  mPromise->MaybeResolve(mOutput);
+
+  mContext->RemoveFromDecodeQueue(this);
+
+}
+
+void
+WebAudioDecodeJob::OnFailure(ErrorCode aErrorCode)
+{
+  MOZ_ASSERT(NS_IsMainThread());
+
+  const char* errorMessage;
+  switch (aErrorCode) {
+  case NoError:
+    MOZ_FALLTHROUGH_ASSERT("Who passed NoError to OnFailure?");
+    // Fall through to get some sort of a sane error message if this actually
+    // happens at runtime.
+  case UnknownError:
+    errorMessage = "MediaDecodeAudioDataUnknownError";
+    break;
+  case UnknownContent:
+    errorMessage = "MediaDecodeAudioDataUnknownContentType";
+    break;
+  case InvalidContent:
+    errorMessage = "MediaDecodeAudioDataInvalidContent";
+    break;
+  case NoAudio:
+    errorMessage = "MediaDecodeAudioDataNoAudio";
+    break;
+  }
+
+  nsIDocument* doc = nullptr;
+  if (nsPIDOMWindowInner* pWindow = mContext->GetParentObject()) {
+    doc = pWindow->GetExtantDoc();
+  }
+  nsContentUtils::ReportToConsole(nsIScriptError::errorFlag,
+                                  NS_LITERAL_CSTRING("Media"),
+                                  doc,
+                                  nsContentUtils::eDOM_PROPERTIES,
+                                  errorMessage);
+
+  // Ignore errors in calling the callback, since there is not much that we can
+  // do about it here.
+  if (mFailureCallback) {
+    mFailureCallback->Call();
+  }
+
+  mPromise->MaybeReject(NS_ERROR_DOM_ENCODING_NOT_SUPPORTED_ERR);
+
+  mContext->RemoveFromDecodeQueue(this);
+}
+
+size_t
+WebAudioDecodeJob::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = 0;
+  amount += mContentType.SizeOfExcludingThisIfUnshared(aMallocSizeOf);
+  if (mSuccessCallback) {
+    amount += mSuccessCallback->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  if (mFailureCallback) {
+    amount += mFailureCallback->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  if (mOutput) {
+    amount += mOutput->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  if (mBuffer) {
+    amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t
+WebAudioDecodeJob::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/MediaBufferDecoder.h b/dom/media/webaudio/MediaBufferDecoder.h
new file mode 100644
index 0000000000..3e79b37ff9
--- /dev/null
+++ b/dom/media/webaudio/MediaBufferDecoder.h
@@ -0,0 +1,79 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaBufferDecoder_h_
+#define MediaBufferDecoder_h_
+
+#include "nsWrapperCache.h"
+#include "nsCOMPtr.h"
+#include "nsString.h"
+#include "nsTArray.h"
+#include "mozilla/dom/TypedArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+
+class ThreadSharedFloatArrayBufferList;
+
+namespace dom {
+class AudioBuffer;
+class AudioContext;
+class DecodeErrorCallback;
+class DecodeSuccessCallback;
+class Promise;
+} // namespace dom
+
+struct WebAudioDecodeJob final
+{
+  // You may omit both the success and failure callback, or you must pass both.
+  // The callbacks are only necessary for asynchronous operation.
+  WebAudioDecodeJob(const nsACString& aContentType,
+                    dom::AudioContext* aContext,
+                    dom::Promise* aPromise,
+                    dom::DecodeSuccessCallback* aSuccessCallback = nullptr,
+                    dom::DecodeErrorCallback* aFailureCallback = nullptr);
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(WebAudioDecodeJob)
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(WebAudioDecodeJob)
+
+  enum ErrorCode {
+    NoError,
+    UnknownContent,
+    UnknownError,
+    InvalidContent,
+    NoAudio
+  };
+
+  typedef void (WebAudioDecodeJob::*ResultFn)(ErrorCode);
+
+  void OnSuccess(ErrorCode /* ignored */);
+  void OnFailure(ErrorCode aErrorCode);
+
+  bool AllocateBuffer();
+
+  size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  nsCString mContentType;
+  uint32_t mWriteIndex;
+  RefPtr<dom::AudioContext> mContext;
+  RefPtr<dom::Promise> mPromise;
+  RefPtr<dom::DecodeSuccessCallback> mSuccessCallback;
+  RefPtr<dom::DecodeErrorCallback> mFailureCallback; // can be null
+  RefPtr<dom::AudioBuffer> mOutput;
+  RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+
+private:
+  ~WebAudioDecodeJob();
+};
+
+void AsyncDecodeWebAudio(const char* aContentType, uint8_t* aBuffer,
+                         uint32_t aLength, WebAudioDecodeJob& aDecodeJob);
+
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/MediaElementAudioSourceNode.cpp b/dom/media/webaudio/MediaElementAudioSourceNode.cpp
new file mode 100644
index 0000000000..ebf7dc44f5
--- /dev/null
+++ b/dom/media/webaudio/MediaElementAudioSourceNode.cpp
@@ -0,0 +1,40 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaElementAudioSourceNode.h"
+#include "mozilla/dom/MediaElementAudioSourceNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+MediaElementAudioSourceNode::MediaElementAudioSourceNode(AudioContext* aContext)
+  : MediaStreamAudioSourceNode(aContext)
+{
+}
+
+/* static */ already_AddRefed<MediaElementAudioSourceNode>
+MediaElementAudioSourceNode::Create(AudioContext* aContext,
+                                    DOMMediaStream* aStream, ErrorResult& aRv)
+{
+  RefPtr<MediaElementAudioSourceNode> node =
+    new MediaElementAudioSourceNode(aContext);
+
+  node->Init(aStream, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  return node.forget();
+}
+
+JSObject*
+MediaElementAudioSourceNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return MediaElementAudioSourceNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/MediaElementAudioSourceNode.h b/dom/media/webaudio/MediaElementAudioSourceNode.h
new file mode 100644
index 0000000000..f6791f355a
--- /dev/null
+++ b/dom/media/webaudio/MediaElementAudioSourceNode.h
@@ -0,0 +1,44 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaElementAudioSourceNode_h_
+#define MediaElementAudioSourceNode_h_
+
+#include "MediaStreamAudioSourceNode.h"
+
+namespace mozilla {
+namespace dom {
+
+class MediaElementAudioSourceNode final : public MediaStreamAudioSourceNode
+{
+public:
+  static already_AddRefed<MediaElementAudioSourceNode>
+  Create(AudioContext* aContext, DOMMediaStream* aStream, ErrorResult& aRv);
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  const char* NodeType() const override
+  {
+    return "MediaElementAudioSourceNode";
+  }
+
+  const char* CrossOriginErrorString() const override
+  {
+    return "MediaElementAudioSourceNodeCrossOrigin";
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+private:
+  explicit MediaElementAudioSourceNode(AudioContext* aContext);
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp b/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp
new file mode 100644
index 0000000000..d8c732e472
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp
@@ -0,0 +1,142 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaStreamAudioDestinationNode.h"
+#include "nsIDocument.h"
+#include "mozilla/dom/MediaStreamAudioDestinationNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "DOMMediaStream.h"
+#include "MediaStreamTrack.h"
+#include "TrackUnionStream.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioDestinationTrackSource :
+  public MediaStreamTrackSource
+{
+public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioDestinationTrackSource,
+                                           MediaStreamTrackSource)
+
+  AudioDestinationTrackSource(MediaStreamAudioDestinationNode* aNode,
+                              nsIPrincipal* aPrincipal)
+    : MediaStreamTrackSource(aPrincipal, nsString())
+    , mNode(aNode)
+  {
+  }
+
+  void Destroy() override
+  {
+    if (mNode) {
+      mNode->DestroyMediaStream();
+      mNode = nullptr;
+    }
+  }
+
+  MediaSourceEnum GetMediaSource() const override
+  {
+    return MediaSourceEnum::AudioCapture;
+  }
+
+  void Stop() override
+  {
+    Destroy();
+  }
+
+private:
+  virtual ~AudioDestinationTrackSource() {}
+
+  RefPtr<MediaStreamAudioDestinationNode> mNode;
+};
+
+NS_IMPL_ADDREF_INHERITED(AudioDestinationTrackSource,
+                         MediaStreamTrackSource)
+NS_IMPL_RELEASE_INHERITED(AudioDestinationTrackSource,
+                          MediaStreamTrackSource)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioDestinationTrackSource)
+NS_INTERFACE_MAP_END_INHERITING(MediaStreamTrackSource)
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioDestinationTrackSource,
+                                   MediaStreamTrackSource,
+                                   mNode)
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(MediaStreamAudioDestinationNode, AudioNode, mDOMStream)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(MediaStreamAudioDestinationNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaStreamAudioDestinationNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(MediaStreamAudioDestinationNode, AudioNode)
+
+MediaStreamAudioDestinationNode::MediaStreamAudioDestinationNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Explicit,
+              ChannelInterpretation::Speakers)
+  , mDOMStream(
+      DOMAudioNodeMediaStream::CreateTrackUnionStreamAsInput(GetOwner(),
+                                                             this,
+                                                             aContext->Graph()))
+{
+  // Ensure an audio track with the correct ID is exposed to JS
+  nsIDocument* doc = aContext->GetParentObject()->GetExtantDoc();
+  RefPtr<MediaStreamTrackSource> source =
+    new AudioDestinationTrackSource(this, doc->NodePrincipal());
+  RefPtr<MediaStreamTrack> track =
+    mDOMStream->CreateDOMTrack(AudioNodeStream::AUDIO_TRACK,
+                               MediaSegment::AUDIO, source,
+                               MediaTrackConstraints());
+  mDOMStream->AddTrackInternal(track);
+
+  ProcessedMediaStream* outputStream = mDOMStream->GetInputStream()->AsProcessedStream();
+  MOZ_ASSERT(!!outputStream);
+  AudioNodeEngine* engine = new AudioNodeEngine(this);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::EXTERNAL_OUTPUT,
+                                    aContext->Graph());
+  mPort = outputStream->AllocateInputPort(mStream, AudioNodeStream::AUDIO_TRACK);
+}
+
+MediaStreamAudioDestinationNode::~MediaStreamAudioDestinationNode()
+{
+}
+
+size_t
+MediaStreamAudioDestinationNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  // Future:
+  // - mDOMStream
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mPort->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+MediaStreamAudioDestinationNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void
+MediaStreamAudioDestinationNode::DestroyMediaStream()
+{
+  AudioNode::DestroyMediaStream();
+  if (mPort) {
+    mPort->Destroy();
+    mPort = nullptr;
+  }
+}
+
+JSObject*
+MediaStreamAudioDestinationNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return MediaStreamAudioDestinationNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/MediaStreamAudioDestinationNode.h b/dom/media/webaudio/MediaStreamAudioDestinationNode.h
new file mode 100644
index 0000000000..6c033b4660
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioDestinationNode.h
@@ -0,0 +1,56 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaStreamAudioDestinationNode_h_
+#define MediaStreamAudioDestinationNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla {
+namespace dom {
+
+class MediaStreamAudioDestinationNode final : public AudioNode
+{
+public:
+  explicit MediaStreamAudioDestinationNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaStreamAudioDestinationNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const final override
+  {
+    return 0;
+  }
+
+  void DestroyMediaStream() override;
+
+  DOMMediaStream* DOMStream() const
+  {
+    return mDOMStream;
+  }
+
+  const char* NodeType() const override
+  {
+    return "MediaStreamAudioDestinationNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~MediaStreamAudioDestinationNode();
+
+private:
+  RefPtr<DOMMediaStream> mDOMStream;
+  RefPtr<MediaInputPort> mPort;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/MediaStreamAudioSourceNode.cpp b/dom/media/webaudio/MediaStreamAudioSourceNode.cpp
new file mode 100644
index 0000000000..beedd53004
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioSourceNode.cpp
@@ -0,0 +1,254 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaStreamAudioSourceNode.h"
+#include "mozilla/dom/MediaStreamAudioSourceNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputStream.h"
+#include "AudioStreamTrack.h"
+#include "nsIDocument.h"
+#include "mozilla/CORSMode.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(MediaStreamAudioSourceNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(MediaStreamAudioSourceNode)
+  tmp->Destroy();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mInputStream)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mInputTrack)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mInputStream)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mInputTrack)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(MediaStreamAudioSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+
+MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+{
+}
+
+/* static */ already_AddRefed<MediaStreamAudioSourceNode>
+MediaStreamAudioSourceNode::Create(AudioContext* aContext,
+                                   DOMMediaStream* aStream, ErrorResult& aRv)
+{
+  RefPtr<MediaStreamAudioSourceNode> node =
+    new MediaStreamAudioSourceNode(aContext);
+
+  node->Init(aStream, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  return node.forget();
+}
+
+void
+MediaStreamAudioSourceNode::Init(DOMMediaStream* aMediaStream, ErrorResult& aRv)
+{
+  if (!aMediaStream) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return;
+  }
+
+  MediaStream* inputStream = aMediaStream->GetPlaybackStream();
+  MediaStreamGraph* graph = Context()->Graph();
+  if (NS_WARN_IF(graph != inputStream->Graph())) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  mInputStream = aMediaStream;
+  AudioNodeEngine* engine = new MediaStreamAudioSourceNodeEngine(this);
+  mStream = AudioNodeExternalInputStream::Create(graph, engine);
+  mInputStream->AddConsumerToKeepAlive(static_cast<nsIDOMEventTarget*>(this));
+
+  mInputStream->RegisterTrackListener(this);
+  AttachToFirstTrack(mInputStream);
+}
+
+void
+MediaStreamAudioSourceNode::Destroy()
+{
+  if (mInputStream) {
+    mInputStream->UnregisterTrackListener(this);
+    mInputStream = nullptr;
+  }
+  DetachFromTrack();
+}
+
+MediaStreamAudioSourceNode::~MediaStreamAudioSourceNode()
+{
+  Destroy();
+}
+
+void
+MediaStreamAudioSourceNode::AttachToTrack(const RefPtr<MediaStreamTrack>& aTrack)
+{
+  MOZ_ASSERT(!mInputTrack);
+  MOZ_ASSERT(aTrack->AsAudioStreamTrack());
+
+  if (!mStream) {
+    return;
+  }
+
+  mInputTrack = aTrack;
+  ProcessedMediaStream* outputStream =
+    static_cast<ProcessedMediaStream*>(mStream.get());
+  mInputPort = mInputTrack->ForwardTrackContentsTo(outputStream);
+  PrincipalChanged(mInputTrack); // trigger enabling/disabling of the connector
+  mInputTrack->AddPrincipalChangeObserver(this);
+}
+
+void
+MediaStreamAudioSourceNode::DetachFromTrack()
+{
+  if (mInputTrack) {
+    mInputTrack->RemovePrincipalChangeObserver(this);
+    mInputTrack = nullptr;
+  }
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+}
+
+void
+MediaStreamAudioSourceNode::AttachToFirstTrack(const RefPtr<DOMMediaStream>& aMediaStream)
+{
+  nsTArray<RefPtr<AudioStreamTrack>> tracks;
+  aMediaStream->GetAudioTracks(tracks);
+
+  for (const RefPtr<AudioStreamTrack>& track : tracks) {
+    if (track->Ended()) {
+      continue;
+    }
+
+    AttachToTrack(track);
+    MarkActive();
+    return;
+  }
+
+  // There was no track available. We'll allow the node to be garbage collected.
+  MarkInactive();
+}
+
+void
+MediaStreamAudioSourceNode::NotifyTrackAdded(const RefPtr<MediaStreamTrack>& aTrack)
+{
+  if (mInputTrack) {
+    return;
+  }
+
+  if (!aTrack->AsAudioStreamTrack()) {
+    return;
+  }
+
+  AttachToTrack(aTrack);
+}
+
+void
+MediaStreamAudioSourceNode::NotifyTrackRemoved(const RefPtr<MediaStreamTrack>& aTrack)
+{
+  if (aTrack != mInputTrack) {
+    return;
+  }
+
+  DetachFromTrack();
+  AttachToFirstTrack(mInputStream);
+}
+
+/**
+ * Changes the principal. Note that this will be called on the main thread, but
+ * changes will be enacted on the MediaStreamGraph thread. If the principal
+ * change results in the document principal losing access to the stream, then
+ * there needs to be other measures in place to ensure that any media that is
+ * governed by the new stream principal is not available to the MediaStreamGraph
+ * before this change completes. Otherwise, a site could get access to
+ * media that they are not authorized to receive.
+ *
+ * One solution is to block the altered content, call this method, then dispatch
+ * another change request to the MediaStreamGraph thread that allows the content
+ * under the new principal to flow. This might be unnecessary if the principal
+ * change is changing to be the document principal.
+ */
+void
+MediaStreamAudioSourceNode::PrincipalChanged(MediaStreamTrack* aMediaStreamTrack)
+{
+  MOZ_ASSERT(aMediaStreamTrack == mInputTrack);
+
+  bool subsumes = false;
+  nsIDocument* doc = nullptr;
+  if (nsPIDOMWindowInner* parent = Context()->GetParentObject()) {
+    doc = parent->GetExtantDoc();
+    if (doc) {
+      nsIPrincipal* docPrincipal = doc->NodePrincipal();
+      nsIPrincipal* trackPrincipal = aMediaStreamTrack->GetPrincipal();
+      if (!trackPrincipal || NS_FAILED(docPrincipal->Subsumes(trackPrincipal, &subsumes))) {
+        subsumes = false;
+      }
+    }
+  }
+  auto stream = static_cast<AudioNodeExternalInputStream*>(mStream.get());
+  bool enabled = subsumes || aMediaStreamTrack->GetCORSMode() != CORS_NONE;
+  stream->SetInt32Parameter(MediaStreamAudioSourceNodeEngine::ENABLE, enabled);
+
+  if (!enabled && doc) {
+    nsContentUtils::ReportToConsole(nsIScriptError::warningFlag,
+                                    NS_LITERAL_CSTRING("Web Audio"),
+                                    doc,
+                                    nsContentUtils::eDOM_PROPERTIES,
+                                    CrossOriginErrorString());
+  }
+}
+
+size_t
+MediaStreamAudioSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  // Future:
+  // - mInputStream
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  if (mInputPort) {
+    amount += mInputPort->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t
+MediaStreamAudioSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void
+MediaStreamAudioSourceNode::DestroyMediaStream()
+{
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+  AudioNode::DestroyMediaStream();
+}
+
+JSObject*
+MediaStreamAudioSourceNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return MediaStreamAudioSourceNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/MediaStreamAudioSourceNode.h b/dom/media/webaudio/MediaStreamAudioSourceNode.h
new file mode 100644
index 0000000000..5383eb2c6a
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioSourceNode.h
@@ -0,0 +1,106 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaStreamAudioSourceNode_h_
+#define MediaStreamAudioSourceNode_h_
+
+#include "AudioNode.h"
+#include "DOMMediaStream.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+
+namespace dom {
+
+class MediaStreamAudioSourceNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit MediaStreamAudioSourceNodeEngine(AudioNode* aNode)
+    : AudioNodeEngine(aNode), mEnabled(false) {}
+
+  bool IsEnabled() const { return mEnabled; }
+  enum Parameters {
+    ENABLE
+  };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override
+  {
+    switch (aIndex) {
+    case ENABLE:
+      mEnabled = !!aValue;
+      break;
+    default:
+      NS_ERROR("MediaStreamAudioSourceNodeEngine bad parameter index");
+    }
+  }
+
+private:
+  bool mEnabled;
+};
+
+class MediaStreamAudioSourceNode : public AudioNode,
+                                   public DOMMediaStream::TrackListener,
+                                   public PrincipalChangeObserver<MediaStreamTrack>
+{
+public:
+  static already_AddRefed<MediaStreamAudioSourceNode>
+  Create(AudioContext* aContext, DOMMediaStream* aStream, ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaStream() override;
+
+  uint16_t NumberOfInputs() const override { return 0; }
+
+  const char* NodeType() const override
+  {
+    return "MediaStreamAudioSourceNode";
+  }
+
+  virtual const char* CrossOriginErrorString() const
+  {
+    return "MediaStreamAudioSourceNodeCrossOrigin";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  // Attaches to aTrack so that its audio content will be used as input.
+  void AttachToTrack(const RefPtr<MediaStreamTrack>& aTrack);
+
+  // Detaches from the currently attached track if there is one.
+  void DetachFromTrack();
+
+  // Attaches to the first available audio track in aMediaStream.
+  void AttachToFirstTrack(const RefPtr<DOMMediaStream>& aMediaStream);
+
+  // From DOMMediaStream::TrackListener.
+  void NotifyTrackAdded(const RefPtr<MediaStreamTrack>& aTrack) override;
+  void NotifyTrackRemoved(const RefPtr<MediaStreamTrack>& aTrack) override;
+
+  // From PrincipalChangeObserver<MediaStreamTrack>.
+  void PrincipalChanged(MediaStreamTrack* aMediaStreamTrack) override;
+
+protected:
+  explicit MediaStreamAudioSourceNode(AudioContext* aContext);
+  void Init(DOMMediaStream* aMediaStream, ErrorResult& aRv);
+  void Destroy();
+  virtual ~MediaStreamAudioSourceNode();
+
+private:
+  RefPtr<MediaInputPort> mInputPort;
+  RefPtr<DOMMediaStream> mInputStream;
+
+  // On construction we set this to the first audio track of mInputStream.
+  RefPtr<MediaStreamTrack> mInputTrack;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/OfflineAudioCompletionEvent.cpp b/dom/media/webaudio/OfflineAudioCompletionEvent.cpp
new file mode 100644
index 0000000000..30a5717193
--- /dev/null
+++ b/dom/media/webaudio/OfflineAudioCompletionEvent.cpp
@@ -0,0 +1,42 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "OfflineAudioCompletionEvent.h"
+#include "mozilla/dom/OfflineAudioCompletionEventBinding.h"
+#include "AudioContext.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(OfflineAudioCompletionEvent, Event,
+                                   mRenderedBuffer)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(OfflineAudioCompletionEvent)
+NS_INTERFACE_MAP_END_INHERITING(Event)
+
+NS_IMPL_ADDREF_INHERITED(OfflineAudioCompletionEvent, Event)
+NS_IMPL_RELEASE_INHERITED(OfflineAudioCompletionEvent, Event)
+
+OfflineAudioCompletionEvent::OfflineAudioCompletionEvent(AudioContext* aOwner,
+                                                         nsPresContext* aPresContext,
+                                                         WidgetEvent* aEvent)
+  : Event(aOwner, aPresContext, aEvent)
+{
+}
+
+OfflineAudioCompletionEvent::~OfflineAudioCompletionEvent()
+{
+}
+
+JSObject*
+OfflineAudioCompletionEvent::WrapObjectInternal(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return OfflineAudioCompletionEventBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/OfflineAudioCompletionEvent.h b/dom/media/webaudio/OfflineAudioCompletionEvent.h
new file mode 100644
index 0000000000..bc21fdec3c
--- /dev/null
+++ b/dom/media/webaudio/OfflineAudioCompletionEvent.h
@@ -0,0 +1,53 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef OfflineAudioCompletionEvent_h_
+#define OfflineAudioCompletionEvent_h_
+
+#include "AudioBuffer.h"
+#include "mozilla/dom/Event.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class OfflineAudioCompletionEvent final : public Event
+{
+public:
+  OfflineAudioCompletionEvent(AudioContext* aOwner,
+                              nsPresContext* aPresContext,
+                              WidgetEvent* aEvent);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_FORWARD_TO_EVENT
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(OfflineAudioCompletionEvent, Event)
+
+  JSObject* WrapObjectInternal(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void InitEvent(AudioBuffer* aRenderedBuffer)
+  {
+    InitEvent(NS_LITERAL_STRING("complete"), false, false);
+    mRenderedBuffer = aRenderedBuffer;
+  }
+
+  AudioBuffer* RenderedBuffer() const
+  {
+    return mRenderedBuffer;
+  }
+
+protected:
+  virtual ~OfflineAudioCompletionEvent();
+
+private:
+  RefPtr<AudioBuffer> mRenderedBuffer;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/OscillatorNode.cpp b/dom/media/webaudio/OscillatorNode.cpp
new file mode 100644
index 0000000000..8e7c103a94
--- /dev/null
+++ b/dom/media/webaudio/OscillatorNode.cpp
@@ -0,0 +1,580 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "OscillatorNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "nsContentUtils.h"
+#include "WebAudioUtils.h"
+#include "blink/PeriodicWave.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(OscillatorNode, AudioNode,
+                                   mPeriodicWave, mFrequency, mDetune)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(OscillatorNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(OscillatorNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(OscillatorNode, AudioNode)
+
+class OscillatorNodeEngine final : public AudioNodeEngine
+{
+public:
+  OscillatorNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mSource(nullptr)
+    , mDestination(aDestination->Stream())
+    , mStart(-1)
+    , mStop(STREAM_TIME_MAX)
+    // Keep the default values in sync with OscillatorNode::OscillatorNode.
+    , mFrequency(440.f)
+    , mDetune(0.f)
+    , mType(OscillatorType::Sine)
+    , mPhase(0.)
+    , mFinalFrequency(0.)
+    , mPhaseIncrement(0.)
+    , mRecomputeParameters(true)
+    , mCustomLength(0)
+    , mCustomDisableNormalization(false)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    mBasicWaveFormCache = aDestination->Context()->GetBasicWaveFormCache();
+  }
+
+  void SetSourceStream(AudioNodeStream* aSource)
+  {
+    mSource = aSource;
+  }
+
+  enum Parameters {
+    FREQUENCY,
+    DETUNE,
+    TYPE,
+    PERIODICWAVE_LENGTH,
+    DISABLE_NORMALIZATION,
+    START,
+    STOP,
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    mRecomputeParameters = true;
+
+    MOZ_ASSERT(mDestination);
+
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case FREQUENCY:
+      mFrequency.InsertEvent<int64_t>(aEvent);
+      break;
+    case DETUNE:
+      mDetune.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad OscillatorNodeEngine TimelineParameter");
+    }
+  }
+
+  void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) override
+  {
+    switch (aIndex) {
+    case START:
+      mStart = aParam;
+      mSource->SetActive();
+      break;
+    case STOP: mStop = aParam; break;
+    default:
+      NS_ERROR("Bad OscillatorNodeEngine StreamTimeParameter");
+    }
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    switch (aIndex) {
+      case TYPE:
+        // Set the new type.
+        mType = static_cast<OscillatorType>(aParam);
+        if (mType == OscillatorType::Sine) {
+          // Forget any previous custom data.
+          mCustomLength = 0;
+          mCustomDisableNormalization = false;
+          mCustom = nullptr;
+          mPeriodicWave = nullptr;
+          mRecomputeParameters = true;
+        }
+        switch (mType) {
+          case OscillatorType::Sine:
+            mPhase = 0.0;
+            break;
+          case OscillatorType::Square:
+          case OscillatorType::Triangle:
+          case OscillatorType::Sawtooth:
+            mPeriodicWave = mBasicWaveFormCache->GetBasicWaveForm(mType);
+            break;
+          case OscillatorType::Custom:
+            break;
+          default:
+            NS_ERROR("Bad OscillatorNodeEngine type parameter.");
+        }
+        // End type switch.
+        break;
+      case PERIODICWAVE_LENGTH:
+        MOZ_ASSERT(aParam >= 0, "negative custom array length");
+        mCustomLength = static_cast<uint32_t>(aParam);
+        break;
+      case DISABLE_NORMALIZATION:
+        MOZ_ASSERT(aParam >= 0, "negative custom array length");
+        mCustomDisableNormalization = static_cast<uint32_t>(aParam);
+        break;
+      default:
+        NS_ERROR("Bad OscillatorNodeEngine Int32Parameter.");
+    }
+    // End index switch.
+  }
+
+  void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer) override
+  {
+    MOZ_ASSERT(mCustomLength, "Custom buffer sent before length");
+    mCustom = aBuffer;
+    MOZ_ASSERT(mCustom->GetChannels() == 2,
+               "PeriodicWave should have sent two channels");
+    mPeriodicWave = WebCore::PeriodicWave::create(mSource->SampleRate(),
+                                                  mCustom->GetData(0),
+                                                  mCustom->GetData(1),
+                                                  mCustomLength,
+                                                  mCustomDisableNormalization);
+  }
+
+  void IncrementPhase()
+  {
+    const float twoPiFloat = float(2 * M_PI);
+    mPhase += mPhaseIncrement;
+    if (mPhase > twoPiFloat) {
+      mPhase -= twoPiFloat;
+    } else if (mPhase < -twoPiFloat) {
+      mPhase += twoPiFloat;
+    }
+  }
+
+  // Returns true if the final frequency (and thus the phase increment) changed,
+  // false otherwise. This allow some optimizations at callsite.
+  bool UpdateParametersIfNeeded(StreamTime ticks, size_t count)
+  {
+    double frequency, detune;
+
+    // Shortcut if frequency-related AudioParam are not automated, and we
+    // already have computed the frequency information and related parameters.
+    if (!ParametersMayNeedUpdate()) {
+      return false;
+    }
+
+    bool simpleFrequency = mFrequency.HasSimpleValue();
+    bool simpleDetune = mDetune.HasSimpleValue();
+
+    if (simpleFrequency) {
+      frequency = mFrequency.GetValue();
+    } else {
+      frequency = mFrequency.GetValueAtTime(ticks, count);
+    }
+    if (simpleDetune) {
+      detune = mDetune.GetValue();
+    } else {
+      detune = mDetune.GetValueAtTime(ticks, count);
+    }
+
+    float finalFrequency = frequency * pow(2., detune / 1200.);
+    float signalPeriod = mSource->SampleRate() / finalFrequency;
+    mRecomputeParameters = false;
+
+    mPhaseIncrement = 2 * M_PI / signalPeriod;
+
+    if (finalFrequency != mFinalFrequency) {
+      mFinalFrequency = finalFrequency;
+      return true;
+    }
+    return false;
+  }
+
+  void FillBounds(float* output, StreamTime ticks,
+                  uint32_t& start, uint32_t& end)
+  {
+    MOZ_ASSERT(output);
+    static_assert(StreamTime(WEBAUDIO_BLOCK_SIZE) < UINT_MAX,
+        "WEBAUDIO_BLOCK_SIZE overflows interator bounds.");
+    start = 0;
+    if (ticks < mStart) {
+      start = mStart - ticks;
+      for (uint32_t i = 0; i < start; ++i) {
+        output[i] = 0.0;
+      }
+    }
+    end = WEBAUDIO_BLOCK_SIZE;
+    if (ticks + end > mStop) {
+      end = mStop - ticks;
+      for (uint32_t i = end; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+        output[i] = 0.0;
+      }
+    }
+  }
+
+  void ComputeSine(float * aOutput, StreamTime ticks, uint32_t aStart, uint32_t aEnd)
+  {
+    for (uint32_t i = aStart; i < aEnd; ++i) {
+      // We ignore the return value, changing the frequency has no impact on
+      // performances here.
+      UpdateParametersIfNeeded(ticks, i);
+
+      aOutput[i] = sin(mPhase);
+
+      IncrementPhase();
+    }
+  }
+
+  bool ParametersMayNeedUpdate()
+  {
+    return !mDetune.HasSimpleValue() ||
+           !mFrequency.HasSimpleValue() ||
+           mRecomputeParameters;
+  }
+
+  void ComputeCustom(float* aOutput,
+                     StreamTime ticks,
+                     uint32_t aStart,
+                     uint32_t aEnd)
+  {
+    MOZ_ASSERT(mPeriodicWave, "No custom waveform data");
+
+    uint32_t periodicWaveSize = mPeriodicWave->periodicWaveSize();
+    // Mask to wrap wave data indices into the range [0,periodicWaveSize).
+    uint32_t indexMask = periodicWaveSize - 1;
+    MOZ_ASSERT(periodicWaveSize && (periodicWaveSize & indexMask) == 0,
+               "periodicWaveSize must be power of 2");
+    float* higherWaveData = nullptr;
+    float* lowerWaveData = nullptr;
+    float tableInterpolationFactor;
+    // Phase increment at frequency of 1 Hz.
+    // mPhase runs [0,periodicWaveSize) here instead of [0,2*M_PI).
+    float basePhaseIncrement = mPeriodicWave->rateScale();
+
+    bool needToFetchWaveData = UpdateParametersIfNeeded(ticks, aStart);
+
+    bool parametersMayNeedUpdate = ParametersMayNeedUpdate();
+    mPeriodicWave->waveDataForFundamentalFrequency(mFinalFrequency,
+                                                   lowerWaveData,
+                                                   higherWaveData,
+                                                   tableInterpolationFactor);
+
+    for (uint32_t i = aStart; i < aEnd; ++i) {
+      if (parametersMayNeedUpdate) {
+        if (needToFetchWaveData) {
+          mPeriodicWave->waveDataForFundamentalFrequency(mFinalFrequency,
+                                                         lowerWaveData,
+                                                         higherWaveData,
+                                                         tableInterpolationFactor);
+        }
+        needToFetchWaveData = UpdateParametersIfNeeded(ticks, i);
+      }
+      // Bilinear interpolation between adjacent samples in each table.
+      float floorPhase = floorf(mPhase);
+      int j1Signed = static_cast<int>(floorPhase);
+      uint32_t j1 = j1Signed & indexMask;
+      uint32_t j2 = j1 + 1;
+      j2 &= indexMask;
+
+      float sampleInterpolationFactor = mPhase - floorPhase;
+
+      float lower = (1.0f - sampleInterpolationFactor) * lowerWaveData[j1] +
+                    sampleInterpolationFactor * lowerWaveData[j2];
+      float higher = (1.0f - sampleInterpolationFactor) * higherWaveData[j1] +
+                    sampleInterpolationFactor * higherWaveData[j2];
+      aOutput[i] = (1.0f - tableInterpolationFactor) * lower +
+                   tableInterpolationFactor * higher;
+
+      // Calculate next phase position from wrapped value j1 to avoid loss of
+      // precision at large values.
+      mPhase =
+        j1 + sampleInterpolationFactor + basePhaseIncrement * mFinalFrequency;
+    }
+  }
+
+  void ComputeSilence(AudioBlock *aOutput)
+  {
+    aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    MOZ_ASSERT(mSource == aStream, "Invalid source stream");
+
+    StreamTime ticks = mDestination->GraphTimeToStreamTime(aFrom);
+    if (mStart == -1) {
+      ComputeSilence(aOutput);
+      return;
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE <= mStart || ticks >= mStop) {
+      ComputeSilence(aOutput);
+
+    } else {
+      aOutput->AllocateChannels(1);
+      float* output = aOutput->ChannelFloatsForWrite(0);
+
+      uint32_t start, end;
+      FillBounds(output, ticks, start, end);
+
+      // Synthesize the correct waveform.
+      switch(mType) {
+        case OscillatorType::Sine:
+          ComputeSine(output, ticks, start, end);
+          break;
+        case OscillatorType::Square:
+        case OscillatorType::Triangle:
+        case OscillatorType::Sawtooth:
+        case OscillatorType::Custom:
+          ComputeCustom(output, ticks, start, end);
+          break;
+        default:
+          ComputeSilence(aOutput);
+      };
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE >= mStop) {
+      // We've finished playing.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // start() has been called.
+    return mStart != -1;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    // Not owned:
+    // - mSource
+    // - mDestination
+    // - mFrequency (internal ref owned by node)
+    // - mDetune (internal ref owned by node)
+
+    if (mCustom) {
+      amount += mCustom->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    if (mPeriodicWave) {
+      amount += mPeriodicWave->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mSource;
+  AudioNodeStream* mDestination;
+  StreamTime mStart;
+  StreamTime mStop;
+  AudioParamTimeline mFrequency;
+  AudioParamTimeline mDetune;
+  OscillatorType mType;
+  float mPhase;
+  float mFinalFrequency;
+  float mPhaseIncrement;
+  bool mRecomputeParameters;
+  RefPtr<ThreadSharedFloatArrayBufferList> mCustom;
+  RefPtr<BasicWaveFormCache> mBasicWaveFormCache;
+  uint32_t mCustomLength;
+  bool mCustomDisableNormalization;
+  RefPtr<WebCore::PeriodicWave> mPeriodicWave;
+};
+
+OscillatorNode::OscillatorNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mType(OscillatorType::Sine)
+  , mFrequency(new AudioParam(this, OscillatorNodeEngine::FREQUENCY,
+                              440.0f, "frequency"))
+  , mDetune(new AudioParam(this, OscillatorNodeEngine::DETUNE, 0.0f, "detune"))
+  , mStartCalled(false)
+{
+  OscillatorNodeEngine* engine = new OscillatorNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NEED_MAIN_THREAD_FINISHED,
+                                    aContext->Graph());
+  engine->SetSourceStream(mStream);
+  mStream->AddMainThreadListener(this);
+}
+
+OscillatorNode::~OscillatorNode()
+{
+}
+
+size_t
+OscillatorNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  // For now only report if we know for sure that it's not shared.
+  if (mPeriodicWave) {
+    amount += mPeriodicWave->SizeOfIncludingThisIfNotShared(aMallocSizeOf);
+  }
+  amount += mFrequency->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+OscillatorNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+OscillatorNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return OscillatorNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+OscillatorNode::DestroyMediaStream()
+{
+  if (mStream) {
+    mStream->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaStream();
+}
+
+void
+OscillatorNode::SendTypeToStream()
+{
+  if (!mStream) {
+    return;
+  }
+  if (mType == OscillatorType::Custom) {
+    // The engine assumes we'll send the custom data before updating the type.
+    SendPeriodicWaveToStream();
+  }
+  SendInt32ParameterToStream(OscillatorNodeEngine::TYPE, static_cast<int32_t>(mType));
+}
+
+void OscillatorNode::SendPeriodicWaveToStream()
+{
+  NS_ASSERTION(mType == OscillatorType::Custom,
+               "Sending custom waveform to engine thread with non-custom type");
+  MOZ_ASSERT(mStream, "Missing node stream.");
+  MOZ_ASSERT(mPeriodicWave, "Send called without PeriodicWave object.");
+  SendInt32ParameterToStream(OscillatorNodeEngine::PERIODICWAVE_LENGTH,
+                             mPeriodicWave->DataLength());
+  SendInt32ParameterToStream(OscillatorNodeEngine::DISABLE_NORMALIZATION,
+                             mPeriodicWave->DisableNormalization());
+  RefPtr<ThreadSharedFloatArrayBufferList> data =
+    mPeriodicWave->GetThreadSharedBuffer();
+  mStream->SetBuffer(data.forget());
+}
+
+void
+OscillatorNode::Start(double aWhen, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+  mStartCalled = true;
+
+  if (!mStream) {
+    // Nothing to play, or we're already dead for some reason
+    return;
+  }
+
+  // TODO: Perhaps we need to do more here.
+  mStream->SetStreamTimeParameter(OscillatorNodeEngine::START,
+                                  Context(), aWhen);
+
+  MarkActive();
+}
+
+void
+OscillatorNode::Stop(double aWhen, ErrorResult& aRv)
+{
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+
+  if (!mStream || !Context()) {
+    // We've already stopped and had our stream shut down
+    return;
+  }
+
+  // TODO: Perhaps we need to do more here.
+  mStream->SetStreamTimeParameter(OscillatorNodeEngine::STOP,
+                                  Context(), std::max(0.0, aWhen));
+}
+
+void
+OscillatorNode::NotifyMainThreadStreamFinished()
+{
+  MOZ_ASSERT(mStream->IsFinished());
+
+  class EndedEventDispatcher final : public Runnable
+  {
+  public:
+    explicit EndedEventDispatcher(OscillatorNode* aNode)
+      : mNode(aNode) {}
+    NS_IMETHOD Run() override
+    {
+      // If it's not safe to run scripts right now, schedule this to run later
+      if (!nsContentUtils::IsSafeToRunScript()) {
+        nsContentUtils::AddScriptRunner(this);
+        return NS_OK;
+      }
+
+      mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
+      // Release stream resources.
+      mNode->DestroyMediaStream();
+      return NS_OK;
+    }
+  private:
+    RefPtr<OscillatorNode> mNode;
+  };
+
+  NS_DispatchToMainThread(new EndedEventDispatcher(this));
+
+  // Drop the playing reference
+  // Warning: The below line might delete this.
+  MarkInactive();
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/OscillatorNode.h b/dom/media/webaudio/OscillatorNode.h
new file mode 100644
index 0000000000..1e17e319e8
--- /dev/null
+++ b/dom/media/webaudio/OscillatorNode.h
@@ -0,0 +1,104 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef OscillatorNode_h_
+#define OscillatorNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "PeriodicWave.h"
+#include "mozilla/dom/OscillatorNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class OscillatorNode final : public AudioNode,
+                             public MainThreadMediaStreamListener
+{
+public:
+  explicit OscillatorNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(OscillatorNode, AudioNode)
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaStream() override;
+
+  uint16_t NumberOfInputs() const final override
+  {
+    return 0;
+  }
+
+  OscillatorType Type() const
+  {
+    return mType;
+  }
+  void SetType(OscillatorType aType, ErrorResult& aRv)
+  {
+    if (aType == OscillatorType::Custom) {
+      // ::Custom can only be set by setPeriodicWave().
+      // https://github.com/WebAudio/web-audio-api/issues/105 for exception.
+      aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+      return;
+    }
+    mType = aType;
+    SendTypeToStream();
+  }
+
+  AudioParam* Frequency() const
+  {
+    return mFrequency;
+  }
+  AudioParam* Detune() const
+  {
+    return mDetune;
+  }
+
+  void Start(double aWhen, ErrorResult& aRv);
+  void Stop(double aWhen, ErrorResult& aRv);
+  void SetPeriodicWave(PeriodicWave& aPeriodicWave)
+  {
+    mPeriodicWave = &aPeriodicWave;
+    // SendTypeToStream will call SendPeriodicWaveToStream for us.
+    mType = OscillatorType::Custom;
+    SendTypeToStream();
+  }
+
+  IMPL_EVENT_HANDLER(ended)
+
+  void NotifyMainThreadStreamFinished() override;
+
+  const char* NodeType() const override
+  {
+    return "OscillatorNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~OscillatorNode();
+
+private:
+  void SendTypeToStream();
+  void SendPeriodicWaveToStream();
+
+private:
+  OscillatorType mType;
+  RefPtr<PeriodicWave> mPeriodicWave;
+  RefPtr<AudioParam> mFrequency;
+  RefPtr<AudioParam> mDetune;
+  bool mStartCalled;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/PannerNode.cpp b/dom/media/webaudio/PannerNode.cpp
new file mode 100644
index 0000000000..7696e984ef
--- /dev/null
+++ b/dom/media/webaudio/PannerNode.cpp
@@ -0,0 +1,786 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "PannerNode.h"
+#include "AlignmentUtils.h"
+#include "AudioDestinationNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioListener.h"
+#include "PanningUtils.h"
+#include "AudioBufferSourceNode.h"
+#include "PlayingRefChangeHandler.h"
+#include "blink/HRTFPanner.h"
+#include "blink/HRTFDatabaseLoader.h"
+#include "nsAutoPtr.h"
+
+using WebCore::HRTFDatabaseLoader;
+using WebCore::HRTFPanner;
+
+namespace mozilla {
+namespace dom {
+
+using namespace std;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(PannerNode)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(PannerNode, AudioNode)
+  if (tmp->Context()) {
+    tmp->Context()->UnregisterPannerNode(tmp);
+  }
+NS_IMPL_CYCLE_COLLECTION_UNLINK(mPositionX, mPositionY, mPositionZ, mOrientationX, mOrientationY, mOrientationZ)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(PannerNode, AudioNode)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPositionX, mPositionY, mPositionZ, mOrientationX, mOrientationY, mOrientationZ)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(PannerNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(PannerNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(PannerNode, AudioNode)
+
+class PannerNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit PannerNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Please keep these default values consistent with PannerNode::PannerNode below.
+    , mPanningModelFunction(&PannerNodeEngine::EqualPowerPanningFunction)
+    , mDistanceModelFunction(&PannerNodeEngine::InverseGainFunction)
+    , mPositionX(0.)
+    , mPositionY(0.)
+    , mPositionZ(0.)
+    , mOrientationX(1.)
+    , mOrientationY(0.)
+    , mOrientationZ(0.)
+    , mVelocity()
+    , mRefDistance(1.)
+    , mMaxDistance(10000.)
+    , mRolloffFactor(1.)
+    , mConeInnerAngle(360.)
+    , mConeOuterAngle(360.)
+    , mConeOuterGain(0.)
+    // These will be initialized when a PannerNode is created, so just initialize them
+    // to some dummy values here.
+    , mListenerDopplerFactor(0.)
+    , mListenerSpeedOfSound(0.)
+    , mLeftOverData(INT_MIN)
+  {
+  }
+
+  void RecvTimelineEvent(uint32_t aIndex, AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+    switch (aIndex) {
+    case PannerNode::POSITIONX:
+      mPositionX.InsertEvent<int64_t>(aEvent);
+      break;
+    case PannerNode::POSITIONY:
+      mPositionY.InsertEvent<int64_t>(aEvent);
+      break;
+    case PannerNode::POSITIONZ:
+      mPositionZ.InsertEvent<int64_t>(aEvent);
+      break;
+    case PannerNode::ORIENTATIONX:
+      mOrientationX.InsertEvent<int64_t>(aEvent);
+      break;
+    case PannerNode::ORIENTATIONY:
+      mOrientationY.InsertEvent<int64_t>(aEvent);
+      break;
+    case PannerNode::ORIENTATIONZ:
+      mOrientationZ.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad PannerNode TimelineParameter");
+    }
+  }
+
+  void CreateHRTFPanner()
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (mHRTFPanner) {
+      return;
+    }
+    // HRTFDatabaseLoader needs to be fetched on the main thread.
+    already_AddRefed<HRTFDatabaseLoader> loader =
+      HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(NodeMainThread()->Context()->SampleRate());
+    mHRTFPanner = new HRTFPanner(NodeMainThread()->Context()->SampleRate(), Move(loader));
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    switch (aIndex) {
+    case PannerNode::PANNING_MODEL:
+      switch (PanningModelType(aParam)) {
+        case PanningModelType::Equalpower:
+          mPanningModelFunction = &PannerNodeEngine::EqualPowerPanningFunction;
+          break;
+        case PanningModelType::HRTF:
+          mPanningModelFunction = &PannerNodeEngine::HRTFPanningFunction;
+          break;
+        default:
+          NS_NOTREACHED("We should never see the alternate names here");
+          break;
+      }
+      break;
+    case PannerNode::DISTANCE_MODEL:
+      switch (DistanceModelType(aParam)) {
+        case DistanceModelType::Inverse:
+          mDistanceModelFunction = &PannerNodeEngine::InverseGainFunction;
+          break;
+        case DistanceModelType::Linear:
+          mDistanceModelFunction = &PannerNodeEngine::LinearGainFunction;
+          break;
+        case DistanceModelType::Exponential:
+          mDistanceModelFunction = &PannerNodeEngine::ExponentialGainFunction;
+          break;
+        default:
+          NS_NOTREACHED("We should never see the alternate names here");
+          break;
+      }
+      break;
+    default:
+      NS_ERROR("Bad PannerNodeEngine Int32Parameter");
+    }
+  }
+  void SetThreeDPointParameter(uint32_t aIndex, const ThreeDPoint& aParam) override
+  {
+    switch (aIndex) {
+    case PannerNode::LISTENER_POSITION: mListenerPosition = aParam; break;
+    case PannerNode::LISTENER_FRONT_VECTOR: mListenerFrontVector = aParam; break;
+    case PannerNode::LISTENER_RIGHT_VECTOR: mListenerRightVector = aParam; break;
+    case PannerNode::LISTENER_VELOCITY: mListenerVelocity = aParam; break;
+    case PannerNode::POSITION:
+      mPositionX.SetValue(aParam.x);
+      mPositionY.SetValue(aParam.y);
+      mPositionZ.SetValue(aParam.z);
+      break;
+    case PannerNode::ORIENTATION:
+      mOrientationX.SetValue(aParam.x);
+      mOrientationY.SetValue(aParam.y);
+      mOrientationZ.SetValue(aParam.z);
+      break;
+    case PannerNode::VELOCITY: mVelocity = aParam; break;
+    default:
+      NS_ERROR("Bad PannerNodeEngine ThreeDPointParameter");
+    }
+  }
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override
+  {
+    switch (aIndex) {
+    case PannerNode::LISTENER_DOPPLER_FACTOR: mListenerDopplerFactor = aParam; break;
+    case PannerNode::LISTENER_SPEED_OF_SOUND: mListenerSpeedOfSound = aParam; break;
+    case PannerNode::REF_DISTANCE: mRefDistance = aParam; break;
+    case PannerNode::MAX_DISTANCE: mMaxDistance = aParam; break;
+    case PannerNode::ROLLOFF_FACTOR: mRolloffFactor = aParam; break;
+    case PannerNode::CONE_INNER_ANGLE: mConeInnerAngle = aParam; break;
+    case PannerNode::CONE_OUTER_ANGLE: mConeOuterAngle = aParam; break;
+    case PannerNode::CONE_OUTER_GAIN: mConeOuterGain = aParam; break;
+    default:
+      NS_ERROR("Bad PannerNodeEngine DoubleParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool *aFinished) override
+  {
+    if (aInput.IsNull()) {
+      // mLeftOverData != INT_MIN means that the panning model was HRTF and a
+      // tail-time reference was added.  Even if the model is now equalpower,
+      // the reference will need to be removed.
+      if (mLeftOverData > 0 &&
+          mPanningModelFunction == &PannerNodeEngine::HRTFPanningFunction) {
+        mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+      } else {
+        if (mLeftOverData != INT_MIN) {
+          mLeftOverData = INT_MIN;
+          aStream->ScheduleCheckForInactive();
+          mHRTFPanner->reset();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::RELEASE);
+          aStream->Graph()->
+            DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+        }
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+      }
+    } else if (mPanningModelFunction == &PannerNodeEngine::HRTFPanningFunction) {
+      if (mLeftOverData == INT_MIN) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+          new PlayingRefChangeHandler(aStream, PlayingRefChangeHandler::ADDREF);
+        aStream->Graph()->
+          DispatchToMainThreadAfterStreamStateUpdate(refchanged.forget());
+      }
+      mLeftOverData = mHRTFPanner->maxTailFrames();
+    }
+
+    StreamTime tick = mDestination->GraphTimeToStreamTime(aFrom);
+    (this->*mPanningModelFunction)(aInput, aOutput, tick);
+  }
+
+  bool IsActive() const override
+  {
+    return mLeftOverData != INT_MIN;
+  }
+
+  void ComputeAzimuthAndElevation(const ThreeDPoint& position, float& aAzimuth, float& aElevation);
+  float ComputeConeGain(const ThreeDPoint& position, const ThreeDPoint& orientation);
+  // Compute how much the distance contributes to the gain reduction.
+  double ComputeDistanceGain(const ThreeDPoint& position);
+
+  void EqualPowerPanningFunction(const AudioBlock& aInput, AudioBlock* aOutput, StreamTime tick);
+  void HRTFPanningFunction(const AudioBlock& aInput, AudioBlock* aOutput, StreamTime tick);
+
+  float LinearGainFunction(double aDistance);
+  float InverseGainFunction(double aDistance);
+  float ExponentialGainFunction(double aDistance);
+
+  ThreeDPoint ConvertAudioParamTimelineTo3DP(AudioParamTimeline& aX, AudioParamTimeline& aY, AudioParamTimeline& aZ, StreamTime& tick);
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    if (mHRTFPanner) {
+      amount += mHRTFPanner->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mDestination;
+  // This member is set on the main thread, but is not accessed on the rendering
+  // thread untile mPanningModelFunction has changed, and this happens strictly
+  // later, via a MediaStreamGraph ControlMessage.
+  nsAutoPtr<HRTFPanner> mHRTFPanner;
+  typedef void (PannerNodeEngine::*PanningModelFunction)(const AudioBlock& aInput, AudioBlock* aOutput, StreamTime tick);
+  PanningModelFunction mPanningModelFunction;
+  typedef float (PannerNodeEngine::*DistanceModelFunction)(double aDistance);
+  DistanceModelFunction mDistanceModelFunction;
+  AudioParamTimeline mPositionX;
+  AudioParamTimeline mPositionY;
+  AudioParamTimeline mPositionZ;
+  AudioParamTimeline mOrientationX;
+  AudioParamTimeline mOrientationY;
+  AudioParamTimeline mOrientationZ;
+  ThreeDPoint mVelocity;
+  double mRefDistance;
+  double mMaxDistance;
+  double mRolloffFactor;
+  double mConeInnerAngle;
+  double mConeOuterAngle;
+  double mConeOuterGain;
+  ThreeDPoint mListenerPosition;
+  ThreeDPoint mListenerFrontVector;
+  ThreeDPoint mListenerRightVector;
+  ThreeDPoint mListenerVelocity;
+  double mListenerDopplerFactor;
+  double mListenerSpeedOfSound;
+  int mLeftOverData;
+};
+
+PannerNode::PannerNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Clamped_max,
+              ChannelInterpretation::Speakers)
+  // Please keep these default values consistent with PannerNodeEngine::PannerNodeEngine above.
+  , mPanningModel(PanningModelType::Equalpower)
+  , mDistanceModel(DistanceModelType::Inverse)
+  , mPositionX(new AudioParam(this, PannerNode::POSITIONX, 0., this->NodeType()))
+  , mPositionY(new AudioParam(this, PannerNode::POSITIONY, 0., this->NodeType()))
+  , mPositionZ(new AudioParam(this, PannerNode::POSITIONZ, 0., this->NodeType()))
+  , mOrientationX(new AudioParam(this, PannerNode::ORIENTATIONX, 1., this->NodeType()))
+  , mOrientationY(new AudioParam(this, PannerNode::ORIENTATIONY, 0., this->NodeType()))
+  , mOrientationZ(new AudioParam(this, PannerNode::ORIENTATIONZ, 0., this->NodeType()))
+  , mVelocity()
+  , mRefDistance(1.)
+  , mMaxDistance(10000.)
+  , mRolloffFactor(1.)
+  , mConeInnerAngle(360.)
+  , mConeOuterAngle(360.)
+  , mConeOuterGain(0.)
+{
+  mStream = AudioNodeStream::Create(aContext,
+                                    new PannerNodeEngine(this, aContext->Destination()),
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+  // We should register once we have set up our stream and engine.
+  Context()->Listener()->RegisterPannerNode(this);
+}
+
+PannerNode::~PannerNode()
+{
+  if (Context()) {
+    Context()->UnregisterPannerNode(this);
+  }
+}
+
+void PannerNode::SetPanningModel(PanningModelType aPanningModel)
+{
+  mPanningModel = aPanningModel;
+  if (mPanningModel == PanningModelType::HRTF) {
+    // We can set the engine's `mHRTFPanner` member here from the main thread,
+    // because the engine will not touch it from the MediaStreamGraph
+    // thread until the PANNING_MODEL message sent below is received.
+    static_cast<PannerNodeEngine*>(mStream->Engine())->CreateHRTFPanner();
+  }
+  SendInt32ParameterToStream(PANNING_MODEL, int32_t(mPanningModel));
+}
+
+size_t
+PannerNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mSources.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+PannerNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+PannerNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return PannerNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void PannerNode::DestroyMediaStream()
+{
+  if (Context()) {
+    Context()->UnregisterPannerNode(this);
+  }
+  AudioNode::DestroyMediaStream();
+}
+
+// Those three functions are described in the spec.
+float
+PannerNodeEngine::LinearGainFunction(double aDistance)
+{
+  return 1 - mRolloffFactor * (std::max(std::min(aDistance, mMaxDistance), mRefDistance) - mRefDistance) / (mMaxDistance - mRefDistance);
+}
+
+float
+PannerNodeEngine::InverseGainFunction(double aDistance)
+{
+  return mRefDistance / (mRefDistance + mRolloffFactor * (std::max(aDistance, mRefDistance) - mRefDistance));
+}
+
+float
+PannerNodeEngine::ExponentialGainFunction(double aDistance)
+{
+  return pow(std::max(aDistance, mRefDistance) / mRefDistance, -mRolloffFactor);
+}
+
+void
+PannerNodeEngine::HRTFPanningFunction(const AudioBlock& aInput,
+                                      AudioBlock* aOutput,
+                                      StreamTime tick)
+{
+  // The output of this node is always stereo, no matter what the inputs are.
+  aOutput->AllocateChannels(2);
+
+  float azimuth, elevation;
+
+  ThreeDPoint position = ConvertAudioParamTimelineTo3DP(mPositionX, mPositionY, mPositionZ, tick);
+  ThreeDPoint orientation = ConvertAudioParamTimelineTo3DP(mOrientationX, mOrientationY, mOrientationZ, tick);
+  if (!orientation.IsZero()) {
+    orientation.Normalize();
+  }
+  ComputeAzimuthAndElevation(position, azimuth, elevation);
+
+  AudioBlock input = aInput;
+  // Gain is applied before the delay and convolution of the HRTF.
+  input.mVolume *= ComputeConeGain(position, orientation) * ComputeDistanceGain(position);
+
+  mHRTFPanner->pan(azimuth, elevation, &input, aOutput);
+}
+
+ThreeDPoint
+PannerNodeEngine::ConvertAudioParamTimelineTo3DP(AudioParamTimeline& aX, AudioParamTimeline& aY, AudioParamTimeline& aZ, StreamTime &tick)
+{
+  return ThreeDPoint(aX.GetValueAtTime(tick),
+                     aY.GetValueAtTime(tick),
+                     aZ.GetValueAtTime(tick));
+}
+
+void
+PannerNodeEngine::EqualPowerPanningFunction(const AudioBlock& aInput,
+                                            AudioBlock* aOutput,
+                                            StreamTime tick)
+{
+  float azimuth, elevation, gainL, gainR, normalizedAzimuth, distanceGain, coneGain;
+  int inputChannels = aInput.ChannelCount();
+
+  // Optimize the case where the position and orientation is constant for this
+  // processing block: we can just apply a constant gain on the left and right
+  // channel
+  if (mPositionX.HasSimpleValue() &&
+      mPositionY.HasSimpleValue() &&
+      mPositionZ.HasSimpleValue() &&
+      mOrientationX.HasSimpleValue() &&
+      mOrientationY.HasSimpleValue() &&
+      mOrientationZ.HasSimpleValue()) {
+
+    ThreeDPoint position = ConvertAudioParamTimelineTo3DP(mPositionX, mPositionY, mPositionZ, tick);
+    ThreeDPoint orientation = ConvertAudioParamTimelineTo3DP(mOrientationX, mOrientationY, mOrientationZ, tick);
+    if (!orientation.IsZero()) {
+      orientation.Normalize();
+    }
+
+    // If both the listener are in the same spot, and no cone gain is specified,
+    // this node is noop.
+    if (mListenerPosition ==  position &&
+        mConeInnerAngle == 360 &&
+        mConeOuterAngle == 360) {
+      *aOutput = aInput;
+      return;
+    }
+
+    // The output of this node is always stereo, no matter what the inputs are.
+    aOutput->AllocateChannels(2);
+
+    ComputeAzimuthAndElevation(position, azimuth, elevation);
+    coneGain = ComputeConeGain(position, orientation);
+
+    // The following algorithm is described in the spec.
+    // Clamp azimuth in the [-90, 90] range.
+    azimuth = min(180.f, max(-180.f, azimuth));
+
+    // Wrap around
+    if (azimuth < -90.f) {
+      azimuth = -180.f - azimuth;
+    } else if (azimuth > 90) {
+      azimuth = 180.f - azimuth;
+    }
+
+    // Normalize the value in the [0, 1] range.
+    if (inputChannels == 1) {
+      normalizedAzimuth = (azimuth + 90.f) / 180.f;
+    } else {
+      if (azimuth <= 0) {
+        normalizedAzimuth = (azimuth + 90.f) / 90.f;
+      } else {
+        normalizedAzimuth = azimuth / 90.f;
+      }
+    }
+
+    distanceGain = ComputeDistanceGain(position);
+
+    // Actually compute the left and right gain.
+    gainL = cos(0.5 * M_PI * normalizedAzimuth);
+    gainR = sin(0.5 * M_PI * normalizedAzimuth);
+
+    // Compute the output.
+    ApplyStereoPanning(aInput, aOutput, gainL, gainR, azimuth <= 0);
+
+    aOutput->mVolume = aInput.mVolume * distanceGain * coneGain;
+  } else {
+    float positionX[WEBAUDIO_BLOCK_SIZE];
+    float positionY[WEBAUDIO_BLOCK_SIZE];
+    float positionZ[WEBAUDIO_BLOCK_SIZE];
+    float orientationX[WEBAUDIO_BLOCK_SIZE];
+    float orientationY[WEBAUDIO_BLOCK_SIZE];
+    float orientationZ[WEBAUDIO_BLOCK_SIZE];
+
+    // The output of this node is always stereo, no matter what the inputs are.
+    aOutput->AllocateChannels(2);
+
+    if (!mPositionX.HasSimpleValue()) {
+      mPositionX.GetValuesAtTime(tick, positionX, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionX[0] = mPositionX.GetValueAtTime(tick);
+    }
+    if (!mPositionY.HasSimpleValue()) {
+      mPositionY.GetValuesAtTime(tick, positionY, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionY[0] = mPositionY.GetValueAtTime(tick);
+    }
+    if (!mPositionZ.HasSimpleValue()) {
+      mPositionZ.GetValuesAtTime(tick, positionZ, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionZ[0] = mPositionZ.GetValueAtTime(tick);
+    }
+    if (!mOrientationX.HasSimpleValue()) {
+      mOrientationX.GetValuesAtTime(tick, orientationX, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationX[0] = mOrientationX.GetValueAtTime(tick);
+    }
+    if (!mOrientationY.HasSimpleValue()) {
+      mOrientationY.GetValuesAtTime(tick, orientationY, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationY[0] = mOrientationY.GetValueAtTime(tick);
+    }
+    if (!mOrientationZ.HasSimpleValue()) {
+      mOrientationZ.GetValuesAtTime(tick, orientationZ, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationZ[0] = mOrientationZ.GetValueAtTime(tick);
+    }
+
+    float computedGain[2*WEBAUDIO_BLOCK_SIZE + 4];
+    bool onLeft[WEBAUDIO_BLOCK_SIZE];
+
+    float* alignedComputedGain = ALIGNED16(computedGain);
+    ASSERT_ALIGNED16(alignedComputedGain);
+    for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+      ThreeDPoint position(mPositionX.HasSimpleValue() ? positionX[0] : positionX[counter],
+                           mPositionY.HasSimpleValue() ? positionY[0] : positionY[counter],
+                           mPositionZ.HasSimpleValue() ? positionZ[0] : positionZ[counter]);
+      ThreeDPoint orientation(mOrientationX.HasSimpleValue() ? orientationX[0] : orientationX[counter],
+                              mOrientationY.HasSimpleValue() ? orientationY[0] : orientationY[counter],
+                              mOrientationZ.HasSimpleValue() ? orientationZ[0] : orientationZ[counter]);
+      if (!orientation.IsZero()) {
+        orientation.Normalize();
+      }
+
+      ComputeAzimuthAndElevation(position, azimuth, elevation);
+      coneGain = ComputeConeGain(position, orientation);
+
+      // The following algorithm is described in the spec.
+      // Clamp azimuth in the [-90, 90] range.
+      azimuth = min(180.f, max(-180.f, azimuth));
+
+      // Wrap around
+      if (azimuth < -90.f) {
+        azimuth = -180.f - azimuth;
+      } else if (azimuth > 90) {
+        azimuth = 180.f - azimuth;
+      }
+
+      // Normalize the value in the [0, 1] range.
+      if (inputChannels == 1) {
+        normalizedAzimuth = (azimuth + 90.f) / 180.f;
+      } else {
+        if (azimuth <= 0) {
+          normalizedAzimuth = (azimuth + 90.f) / 90.f;
+        } else {
+          normalizedAzimuth = azimuth / 90.f;
+        }
+      }
+
+      distanceGain = ComputeDistanceGain(position);
+
+      // Actually compute the left and right gain.
+      float gainL = cos(0.5 * M_PI * normalizedAzimuth) * aInput.mVolume * distanceGain * coneGain;
+      float gainR = sin(0.5 * M_PI * normalizedAzimuth) * aInput.mVolume * distanceGain * coneGain;
+
+      alignedComputedGain[counter] = gainL;
+      alignedComputedGain[WEBAUDIO_BLOCK_SIZE + counter] = gainR;
+      onLeft[counter] = azimuth <= 0;
+    }
+
+    // Apply the gain to the output buffer
+    ApplyStereoPanning(aInput, aOutput, alignedComputedGain, &alignedComputedGain[WEBAUDIO_BLOCK_SIZE], onLeft);
+
+  }
+}
+
+// This algorithm is specified in the webaudio spec.
+void
+PannerNodeEngine::ComputeAzimuthAndElevation(const ThreeDPoint& position, float& aAzimuth, float& aElevation)
+{
+  ThreeDPoint sourceListener = position - mListenerPosition;
+  if (sourceListener.IsZero()) {
+    aAzimuth = 0.0;
+    aElevation = 0.0;
+    return;
+  }
+
+  sourceListener.Normalize();
+
+  // Project the source-listener vector on the x-z plane.
+  const ThreeDPoint& listenerFront = mListenerFrontVector;
+  const ThreeDPoint& listenerRight = mListenerRightVector;
+  ThreeDPoint up = listenerRight.CrossProduct(listenerFront);
+
+  double upProjection = sourceListener.DotProduct(up);
+  aElevation = 90 - 180 * acos(upProjection) / M_PI;
+
+  if (aElevation > 90) {
+    aElevation = 180 - aElevation;
+  } else if (aElevation < -90) {
+    aElevation = -180 - aElevation;
+  }
+
+  ThreeDPoint projectedSource = sourceListener - up * upProjection;
+  if (projectedSource.IsZero()) {
+    // source - listener direction is up or down.
+    aAzimuth = 0.0;
+    return;
+  }
+  projectedSource.Normalize();
+
+  // Actually compute the angle, and convert to degrees
+  double projection = projectedSource.DotProduct(listenerRight);
+  aAzimuth = 180 * acos(projection) / M_PI;
+
+  // Compute whether the source is in front or behind the listener.
+  double frontBack = projectedSource.DotProduct(listenerFront);
+  if (frontBack < 0) {
+    aAzimuth = 360 - aAzimuth;
+  }
+  // Rotate the azimuth so it is relative to the listener front vector instead
+  // of the right vector.
+  if ((aAzimuth >= 0) && (aAzimuth <= 270)) {
+    aAzimuth = 90 - aAzimuth;
+  } else {
+    aAzimuth = 450 - aAzimuth;
+  }
+}
+
+// This algorithm is described in the WebAudio spec.
+float
+PannerNodeEngine::ComputeConeGain(const ThreeDPoint& position,
+                                  const ThreeDPoint& orientation)
+{
+  // Omnidirectional source
+  if (orientation.IsZero() || ((mConeInnerAngle == 360) && (mConeOuterAngle == 360))) {
+    return 1;
+  }
+
+  // Normalized source-listener vector
+  ThreeDPoint sourceToListener = mListenerPosition - position;
+  sourceToListener.Normalize();
+
+  // Angle between the source orientation vector and the source-listener vector
+  double dotProduct = sourceToListener.DotProduct(orientation);
+  double angle = 180 * acos(dotProduct) / M_PI;
+  double absAngle = fabs(angle);
+
+  // Divide by 2 here since API is entire angle (not half-angle)
+  double absInnerAngle = fabs(mConeInnerAngle) / 2;
+  double absOuterAngle = fabs(mConeOuterAngle) / 2;
+  double gain = 1;
+
+  if (absAngle <= absInnerAngle) {
+    // No attenuation
+    gain = 1;
+  } else if (absAngle >= absOuterAngle) {
+    // Max attenuation
+    gain = mConeOuterGain;
+  } else {
+    // Between inner and outer cones
+    // inner -> outer, x goes from 0 -> 1
+    double x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);
+    gain = (1 - x) + mConeOuterGain * x;
+  }
+
+  return gain;
+}
+
+double
+PannerNodeEngine::ComputeDistanceGain(const ThreeDPoint& position)
+{
+  ThreeDPoint distanceVec = position - mListenerPosition;
+  float distance = sqrt(distanceVec.DotProduct(distanceVec));
+  return std::max(0.0f, (this->*mDistanceModelFunction)(distance));
+}
+
+float
+PannerNode::ComputeDopplerShift()
+{
+  double dopplerShift = 1.0; // Initialize to default value
+
+  AudioListener* listener = Context()->Listener();
+
+  if (listener->DopplerFactor() > 0) {
+    // Don't bother if both source and listener have no velocity.
+    if (!mVelocity.IsZero() || !listener->Velocity().IsZero()) {
+      // Calculate the source to listener vector.
+      ThreeDPoint sourceToListener = ConvertAudioParamTo3DP(mPositionX, mPositionY, mPositionZ) - listener->Velocity();
+
+      double sourceListenerMagnitude = sourceToListener.Magnitude();
+
+      double listenerProjection = sourceToListener.DotProduct(listener->Velocity()) / sourceListenerMagnitude;
+      double sourceProjection = sourceToListener.DotProduct(mVelocity) / sourceListenerMagnitude;
+
+      listenerProjection = -listenerProjection;
+      sourceProjection = -sourceProjection;
+
+      double scaledSpeedOfSound = listener->SpeedOfSound() / listener->DopplerFactor();
+      listenerProjection = min(listenerProjection, scaledSpeedOfSound);
+      sourceProjection = min(sourceProjection, scaledSpeedOfSound);
+
+      dopplerShift = ((listener->SpeedOfSound() - listener->DopplerFactor() * listenerProjection) / (listener->SpeedOfSound() - listener->DopplerFactor() * sourceProjection));
+
+      WebAudioUtils::FixNaN(dopplerShift); // Avoid illegal values
+
+      // Limit the pitch shifting to 4 octaves up and 3 octaves down.
+      dopplerShift = min(dopplerShift, 16.);
+      dopplerShift = max(dopplerShift, 0.125);
+    }
+  }
+
+  return dopplerShift;
+}
+
+void
+PannerNode::FindConnectedSources()
+{
+  mSources.Clear();
+  std::set<AudioNode*> cycleSet;
+  FindConnectedSources(this, mSources, cycleSet);
+}
+
+void
+PannerNode::FindConnectedSources(AudioNode* aNode,
+                                 nsTArray<AudioBufferSourceNode*>& aSources,
+                                 std::set<AudioNode*>& aNodesSeen)
+{
+  if (!aNode) {
+    return;
+  }
+
+  const nsTArray<InputNode>& inputNodes = aNode->InputNodes();
+
+  for(unsigned i = 0; i < inputNodes.Length(); i++) {
+    // Return if we find a node that we have seen already.
+    if (aNodesSeen.find(inputNodes[i].mInputNode) != aNodesSeen.end()) {
+      return;
+    }
+    aNodesSeen.insert(inputNodes[i].mInputNode);
+    // Recurse
+    FindConnectedSources(inputNodes[i].mInputNode, aSources, aNodesSeen);
+
+    // Check if this node is an AudioBufferSourceNode that still have a stream,
+    // which means it has not finished playing.
+    AudioBufferSourceNode* node = inputNodes[i].mInputNode->AsAudioBufferSourceNode();
+    if (node && node->GetStream()) {
+      aSources.AppendElement(node);
+    }
+  }
+}
+
+void
+PannerNode::SendDopplerToSourcesIfNeeded()
+{
+  // Don't bother sending the doppler shift if both the source and the listener
+  // are not moving, because the doppler shift is going to be 1.0.
+  if (!(Context()->Listener()->Velocity().IsZero() && mVelocity.IsZero())) {
+    for(uint32_t i = 0; i < mSources.Length(); i++) {
+      mSources[i]->SendDopplerShiftToStream(ComputeDopplerShift());
+    }
+  }
+}
+
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/PannerNode.h b/dom/media/webaudio/PannerNode.h
new file mode 100644
index 0000000000..184db46036
--- /dev/null
+++ b/dom/media/webaudio/PannerNode.h
@@ -0,0 +1,296 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PannerNode_h_
+#define PannerNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/PannerNodeBinding.h"
+#include "ThreeDPoint.h"
+#include "mozilla/WeakPtr.h"
+#include <limits>
+#include <set>
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+class AudioBufferSourceNode;
+
+class PannerNode final : public AudioNode,
+                         public SupportsWeakPtr<PannerNode>
+{
+public:
+  MOZ_DECLARE_WEAKREFERENCE_TYPENAME(PannerNode)
+  explicit PannerNode(AudioContext* aContext);
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaStream() override;
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override
+  {
+    if (aChannelCount > 2) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode, ErrorResult& aRv) override
+  {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(PannerNode, AudioNode)
+
+  PanningModelType PanningModel() const
+  {
+    return mPanningModel;
+  }
+  void SetPanningModel(PanningModelType aPanningModel);
+
+  DistanceModelType DistanceModel() const
+  {
+    return mDistanceModel;
+  }
+  void SetDistanceModel(DistanceModelType aDistanceModel)
+  {
+    mDistanceModel = aDistanceModel;
+    SendInt32ParameterToStream(DISTANCE_MODEL, int32_t(mDistanceModel));
+  }
+
+  void SetPosition(double aX, double aY, double aZ)
+  {
+    if (fabs(aX) > std::numeric_limits<float>::max() ||
+        fabs(aY) > std::numeric_limits<float>::max() ||
+        fabs(aZ) > std::numeric_limits<float>::max()) {
+      return;
+    }
+    mPositionX->SetValue(aX);
+    mPositionY->SetValue(aY);
+    mPositionZ->SetValue(aZ);
+    SendThreeDPointParameterToStream(POSITION, ConvertAudioParamTo3DP(mPositionX, mPositionY, mPositionZ));
+  }
+
+  void SetOrientation(double aX, double aY, double aZ)
+  {
+    if (fabs(aX) > std::numeric_limits<float>::max() ||
+        fabs(aY) > std::numeric_limits<float>::max() ||
+        fabs(aZ) > std::numeric_limits<float>::max()) {
+      return;
+    }
+    mOrientationX->SetValue(aX);
+    mOrientationY->SetValue(aY);
+    mOrientationZ->SetValue(aZ);
+    SendThreeDPointParameterToStream(ORIENTATION, ConvertAudioParamTo3DP(mOrientationX, mOrientationY, mOrientationZ));
+  }
+
+  void SetVelocity(double aX, double aY, double aZ)
+  {
+    if (WebAudioUtils::FuzzyEqual(mVelocity.x, aX) &&
+        WebAudioUtils::FuzzyEqual(mVelocity.y, aY) &&
+        WebAudioUtils::FuzzyEqual(mVelocity.z, aZ)) {
+      return;
+    }
+    mVelocity.x = aX;
+    mVelocity.y = aY;
+    mVelocity.z = aZ;
+    SendThreeDPointParameterToStream(VELOCITY, mVelocity);
+    SendDopplerToSourcesIfNeeded();
+  }
+
+  double RefDistance() const
+  {
+    return mRefDistance;
+  }
+  void SetRefDistance(double aRefDistance)
+  {
+    if (WebAudioUtils::FuzzyEqual(mRefDistance, aRefDistance)) {
+      return;
+    }
+    mRefDistance = aRefDistance;
+    SendDoubleParameterToStream(REF_DISTANCE, mRefDistance);
+  }
+
+  double MaxDistance() const
+  {
+    return mMaxDistance;
+  }
+  void SetMaxDistance(double aMaxDistance)
+  {
+    if (WebAudioUtils::FuzzyEqual(mMaxDistance, aMaxDistance)) {
+      return;
+    }
+    mMaxDistance = aMaxDistance;
+    SendDoubleParameterToStream(MAX_DISTANCE, mMaxDistance);
+  }
+
+  double RolloffFactor() const
+  {
+    return mRolloffFactor;
+  }
+  void SetRolloffFactor(double aRolloffFactor)
+  {
+    if (WebAudioUtils::FuzzyEqual(mRolloffFactor, aRolloffFactor)) {
+      return;
+    }
+    mRolloffFactor = aRolloffFactor;
+    SendDoubleParameterToStream(ROLLOFF_FACTOR, mRolloffFactor);
+  }
+
+  double ConeInnerAngle() const
+  {
+    return mConeInnerAngle;
+  }
+  void SetConeInnerAngle(double aConeInnerAngle)
+  {
+    if (WebAudioUtils::FuzzyEqual(mConeInnerAngle, aConeInnerAngle)) {
+      return;
+    }
+    mConeInnerAngle = aConeInnerAngle;
+    SendDoubleParameterToStream(CONE_INNER_ANGLE, mConeInnerAngle);
+  }
+
+  double ConeOuterAngle() const
+  {
+    return mConeOuterAngle;
+  }
+  void SetConeOuterAngle(double aConeOuterAngle)
+  {
+    if (WebAudioUtils::FuzzyEqual(mConeOuterAngle, aConeOuterAngle)) {
+      return;
+    }
+    mConeOuterAngle = aConeOuterAngle;
+    SendDoubleParameterToStream(CONE_OUTER_ANGLE, mConeOuterAngle);
+  }
+
+  double ConeOuterGain() const
+  {
+    return mConeOuterGain;
+  }
+  void SetConeOuterGain(double aConeOuterGain)
+  {
+    if (WebAudioUtils::FuzzyEqual(mConeOuterGain, aConeOuterGain)) {
+      return;
+    }
+    mConeOuterGain = aConeOuterGain;
+    SendDoubleParameterToStream(CONE_OUTER_GAIN, mConeOuterGain);
+  }
+
+  AudioParam* PositionX()
+  {
+    return mPositionX;
+  }
+
+  AudioParam* PositionY()
+  {
+    return mPositionY;
+  }
+
+  AudioParam* PositionZ()
+  {
+    return mPositionZ;
+  }
+
+  AudioParam* OrientationX()
+  {
+    return mOrientationX;
+  }
+
+  AudioParam* OrientationY()
+  {
+    return mOrientationY;
+  }
+
+  AudioParam* OrientationZ()
+  {
+    return mOrientationZ;
+  }
+
+
+  float ComputeDopplerShift();
+  void SendDopplerToSourcesIfNeeded();
+  void FindConnectedSources();
+  void FindConnectedSources(AudioNode* aNode, nsTArray<AudioBufferSourceNode*>& aSources, std::set<AudioNode*>& aSeenNodes);
+
+  const char* NodeType() const override
+  {
+    return "PannerNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~PannerNode();
+
+private:
+  friend class AudioListener;
+  friend class PannerNodeEngine;
+  enum EngineParameters {
+    LISTENER_POSITION,
+    LISTENER_FRONT_VECTOR, // unit length
+    LISTENER_RIGHT_VECTOR, // unit length, orthogonal to LISTENER_FRONT_VECTOR
+    LISTENER_VELOCITY,
+    LISTENER_DOPPLER_FACTOR,
+    LISTENER_SPEED_OF_SOUND,
+    PANNING_MODEL,
+    DISTANCE_MODEL,
+    POSITION,
+    POSITIONX,
+    POSITIONY,
+    POSITIONZ,
+    ORIENTATION, // unit length or zero
+    ORIENTATIONX,
+    ORIENTATIONY,
+    ORIENTATIONZ,
+    VELOCITY,
+    REF_DISTANCE,
+    MAX_DISTANCE,
+    ROLLOFF_FACTOR,
+    CONE_INNER_ANGLE,
+    CONE_OUTER_ANGLE,
+    CONE_OUTER_GAIN
+  };
+
+  ThreeDPoint ConvertAudioParamTo3DP(RefPtr <AudioParam> aX, RefPtr <AudioParam> aY, RefPtr <AudioParam> aZ)
+  {
+    return ThreeDPoint(aX->GetValue(), aY->GetValue(), aZ->GetValue());
+  }
+
+  PanningModelType mPanningModel;
+  DistanceModelType mDistanceModel;
+  RefPtr<AudioParam> mPositionX;
+  RefPtr<AudioParam> mPositionY;
+  RefPtr<AudioParam> mPositionZ;
+  RefPtr<AudioParam> mOrientationX;
+  RefPtr<AudioParam> mOrientationY;
+  RefPtr<AudioParam> mOrientationZ;
+  ThreeDPoint mVelocity;
+
+  double mRefDistance;
+  double mMaxDistance;
+  double mRolloffFactor;
+  double mConeInnerAngle;
+  double mConeOuterAngle;
+  double mConeOuterGain;
+
+  // An array of all the AudioBufferSourceNode connected directly or indirectly
+  // to this AudioPannerNode.
+  nsTArray<AudioBufferSourceNode*> mSources;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/PanningUtils.h b/dom/media/webaudio/PanningUtils.h
new file mode 100644
index 0000000000..a3be3f45e0
--- /dev/null
+++ b/dom/media/webaudio/PanningUtils.h
@@ -0,0 +1,65 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PANNING_UTILS_H
+#define PANNING_UTILS_H
+
+#include "AudioSegment.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+namespace dom {
+
+template<typename T>
+void
+GainMonoToStereo(const AudioBlock& aInput, AudioBlock* aOutput,
+                 T aGainL, T aGainR)
+{
+  float* outputL = aOutput->ChannelFloatsForWrite(0);
+  float* outputR = aOutput->ChannelFloatsForWrite(1);
+  const float* input = static_cast<const float*>(aInput.mChannelData[0]);
+
+  MOZ_ASSERT(aInput.ChannelCount() == 1);
+  MOZ_ASSERT(aOutput->ChannelCount() == 2);
+
+  AudioBlockPanMonoToStereo(input, aGainL, aGainR, outputL, outputR);
+}
+
+// T can be float or an array of float, and  U can be bool or an array of bool,
+// depending if the value of the parameters are constant for this block.
+template<typename T, typename U>
+void
+GainStereoToStereo(const AudioBlock& aInput, AudioBlock* aOutput,
+                   T aGainL, T aGainR, U aOnLeft)
+{
+  float* outputL = aOutput->ChannelFloatsForWrite(0);
+  float* outputR = aOutput->ChannelFloatsForWrite(1);
+  const float* inputL = static_cast<const float*>(aInput.mChannelData[0]);
+  const float* inputR = static_cast<const float*>(aInput.mChannelData[1]);
+
+  MOZ_ASSERT(aInput.ChannelCount() == 2);
+  MOZ_ASSERT(aOutput->ChannelCount() == 2);
+
+  AudioBlockPanStereoToStereo(inputL, inputR, aGainL, aGainR, aOnLeft, outputL, outputR);
+}
+
+// T can be float or an array of float, and  U can be bool or an array of bool,
+// depending if the value of the parameters are constant for this block.
+template<typename T, typename U>
+void ApplyStereoPanning(const AudioBlock& aInput, AudioBlock* aOutput,
+                        T aGainL, T aGainR, U aOnLeft)
+{
+  if (aInput.ChannelCount() == 1) {
+    GainMonoToStereo(aInput, aOutput, aGainL, aGainR);
+  } else {
+    GainStereoToStereo(aInput, aOutput, aGainL, aGainR, aOnLeft);
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
+
+#endif // PANNING_UTILS_H
diff --git a/dom/media/webaudio/PeriodicWave.cpp b/dom/media/webaudio/PeriodicWave.cpp
new file mode 100644
index 0000000000..396a93e138
--- /dev/null
+++ b/dom/media/webaudio/PeriodicWave.cpp
@@ -0,0 +1,74 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "PeriodicWave.h"
+#include "AudioContext.h"
+#include "mozilla/dom/PeriodicWaveBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(PeriodicWave, mContext)
+
+NS_IMPL_CYCLE_COLLECTION_ROOT_NATIVE(PeriodicWave, AddRef)
+NS_IMPL_CYCLE_COLLECTION_UNROOT_NATIVE(PeriodicWave, Release)
+
+PeriodicWave::PeriodicWave(AudioContext* aContext,
+                           const float* aRealData,
+                           const float* aImagData,
+                           const uint32_t aLength,
+                           const bool aDisableNormalization,
+                           ErrorResult& aRv)
+  : mContext(aContext)
+  , mDisableNormalization(aDisableNormalization)
+{
+  MOZ_ASSERT(aContext);
+
+  // Caller should have checked this and thrown.
+  MOZ_ASSERT(aLength > 0);
+  mLength = aLength;
+
+  // Copy coefficient data. The two arrays share an allocation.
+  mCoefficients = new ThreadSharedFloatArrayBufferList(2);
+  float* buffer = static_cast<float*>(malloc(aLength*sizeof(float)*2));
+  if (buffer == nullptr) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+  PodCopy(buffer, aRealData, aLength);
+  mCoefficients->SetData(0, buffer, free, buffer);
+  PodCopy(buffer+aLength, aImagData, aLength);
+  mCoefficients->SetData(1, nullptr, free, buffer+aLength);
+}
+
+size_t
+PeriodicWave::SizeOfExcludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const
+{
+  // Not owned:
+  // - mContext
+  size_t amount = 0;
+  if (!mCoefficients->IsShared()) {
+    amount += mCoefficients->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t
+PeriodicWave::SizeOfIncludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThisIfNotShared(aMallocSizeOf);
+}
+
+JSObject*
+PeriodicWave::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return PeriodicWaveBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/PeriodicWave.h b/dom/media/webaudio/PeriodicWave.h
new file mode 100644
index 0000000000..b67d597e4d
--- /dev/null
+++ b/dom/media/webaudio/PeriodicWave.h
@@ -0,0 +1,70 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PeriodicWave_h_
+#define PeriodicWave_h_
+
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "AudioContext.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+
+namespace dom {
+
+class PeriodicWave final : public nsWrapperCache
+{
+public:
+  PeriodicWave(AudioContext* aContext,
+               const float* aRealData,
+               const float* aImagData,
+               const uint32_t aLength,
+               const bool aDisableNormalization,
+               ErrorResult& aRv);
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(PeriodicWave)
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(PeriodicWave)
+
+  AudioContext* GetParentObject() const
+  {
+    return mContext;
+  }
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  uint32_t DataLength() const
+  {
+    return mLength;
+  }
+
+  bool DisableNormalization() const
+  {
+    return mDisableNormalization;
+  }
+
+  ThreadSharedFloatArrayBufferList* GetThreadSharedBuffer() const
+  {
+    return mCoefficients;
+  }
+
+  size_t SizeOfExcludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const;
+  size_t SizeOfIncludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const;
+
+private:
+  ~PeriodicWave() {}
+
+  RefPtr<AudioContext> mContext;
+  RefPtr<ThreadSharedFloatArrayBufferList> mCoefficients;
+  uint32_t mLength;
+  bool mDisableNormalization;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/PlayingRefChangeHandler.h b/dom/media/webaudio/PlayingRefChangeHandler.h
new file mode 100644
index 0000000000..6436d1dbc3
--- /dev/null
+++ b/dom/media/webaudio/PlayingRefChangeHandler.h
@@ -0,0 +1,48 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PlayingRefChangeHandler_h__
+#define PlayingRefChangeHandler_h__
+
+#include "nsThreadUtils.h"
+#include "AudioNodeStream.h"
+
+namespace mozilla {
+namespace dom {
+
+class PlayingRefChangeHandler final : public Runnable
+{
+public:
+  enum ChangeType { ADDREF, RELEASE };
+  PlayingRefChangeHandler(AudioNodeStream* aStream, ChangeType aChange)
+    : mStream(aStream)
+    , mChange(aChange)
+  {
+  }
+
+  NS_IMETHOD Run() override
+  {
+    RefPtr<AudioNode> node = mStream->Engine()->NodeMainThread();
+    if (node) {
+      if (mChange == ADDREF) {
+        node->MarkActive();
+      } else if (mChange == RELEASE) {
+        node->MarkInactive();
+      }
+    }
+    return NS_OK;
+  }
+
+private:
+  RefPtr<AudioNodeStream> mStream;
+  ChangeType mChange;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/ReportDecodeResultTask.h b/dom/media/webaudio/ReportDecodeResultTask.h
new file mode 100644
index 0000000000..5d34f3438c
--- /dev/null
+++ b/dom/media/webaudio/ReportDecodeResultTask.h
@@ -0,0 +1,43 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ReportDecodeResultTask_h_
+#define ReportDecodeResultTask_h_
+
+#include "mozilla/Attributes.h"
+#include "MediaBufferDecoder.h"
+
+namespace mozilla {
+
+class ReportDecodeResultTask final : public Runnable
+{
+public:
+  ReportDecodeResultTask(DecodeJob& aDecodeJob,
+                         DecodeJob::ResultFn aFunction)
+    : mDecodeJob(aDecodeJob)
+    , mFunction(aFunction)
+  {
+    MOZ_ASSERT(aFunction);
+  }
+
+  NS_IMETHOD Run() override
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    (mDecodeJob.*mFunction)();
+
+    return NS_OK;
+  }
+
+private:
+  DecodeJob& mDecodeJob;
+  DecodeJob::ResultFn mFunction;
+};
+
+}
+
+#endif
+
diff --git a/dom/media/webaudio/ScriptProcessorNode.cpp b/dom/media/webaudio/ScriptProcessorNode.cpp
new file mode 100644
index 0000000000..3b5df51eff
--- /dev/null
+++ b/dom/media/webaudio/ScriptProcessorNode.cpp
@@ -0,0 +1,573 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "ScriptProcessorNode.h"
+#include "mozilla/dom/ScriptProcessorNodeBinding.h"
+#include "AudioBuffer.h"
+#include "AudioDestinationNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioProcessingEvent.h"
+#include "WebAudioUtils.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/PodOperations.h"
+#include "nsAutoPtr.h"
+#include <deque>
+
+namespace mozilla {
+namespace dom {
+
+// The maximum latency, in seconds, that we can live with before dropping
+// buffers.
+static const float MAX_LATENCY_S = 0.5;
+
+NS_IMPL_ISUPPORTS_INHERITED0(ScriptProcessorNode, AudioNode)
+
+// This class manages a queue of output buffers shared between
+// the main thread and the Media Stream Graph thread.
+class SharedBuffers final
+{
+private:
+  class OutputQueue final
+  {
+  public:
+    explicit OutputQueue(const char* aName)
+      : mMutex(aName)
+    {}
+
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+    {
+      mMutex.AssertCurrentThreadOwns();
+
+      size_t amount = 0;
+      for (size_t i = 0; i < mBufferList.size(); i++) {
+        amount += mBufferList[i].SizeOfExcludingThis(aMallocSizeOf, false);
+      }
+
+      return amount;
+    }
+
+    Mutex& Lock() const { return const_cast<OutputQueue*>(this)->mMutex; }
+
+    size_t ReadyToConsume() const
+    {
+      // Accessed on both main thread and media graph thread.
+      mMutex.AssertCurrentThreadOwns();
+      return mBufferList.size();
+    }
+
+    // Produce one buffer
+    AudioChunk& Produce()
+    {
+      mMutex.AssertCurrentThreadOwns();
+      MOZ_ASSERT(NS_IsMainThread());
+      mBufferList.push_back(AudioChunk());
+      return mBufferList.back();
+    }
+
+    // Consumes one buffer.
+    AudioChunk Consume()
+    {
+      mMutex.AssertCurrentThreadOwns();
+      MOZ_ASSERT(!NS_IsMainThread());
+      MOZ_ASSERT(ReadyToConsume() > 0);
+      AudioChunk front = mBufferList.front();
+      mBufferList.pop_front();
+      return front;
+    }
+
+    // Empties the buffer queue.
+    void Clear()
+    {
+      mMutex.AssertCurrentThreadOwns();
+      mBufferList.clear();
+    }
+
+  private:
+    typedef std::deque<AudioChunk> BufferList;
+
+    // Synchronizes access to mBufferList.  Note that it's the responsibility
+    // of the callers to perform the required locking, and we assert that every
+    // time we access mBufferList.
+    Mutex mMutex;
+    // The list representing the queue.
+    BufferList mBufferList;
+  };
+
+public:
+  explicit SharedBuffers(float aSampleRate)
+    : mOutputQueue("SharedBuffers::outputQueue")
+    , mDelaySoFar(STREAM_TIME_MAX)
+    , mSampleRate(aSampleRate)
+    , mLatency(0.0)
+    , mDroppingBuffers(false)
+  {
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    size_t amount = aMallocSizeOf(this);
+
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      amount += mOutputQueue.SizeOfExcludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  // main thread
+  void FinishProducingOutputBuffer(ThreadSharedFloatArrayBufferList* aBuffer,
+                                   uint32_t aBufferSize)
+  {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    TimeStamp now = TimeStamp::Now();
+
+    if (mLastEventTime.IsNull()) {
+      mLastEventTime = now;
+    } else {
+      // When main thread blocking has built up enough so
+      // |mLatency > MAX_LATENCY_S|, frame dropping starts. It continues until
+      // the output buffer is completely empty, at which point the accumulated
+      // latency is also reset to 0.
+      // It could happen that the output queue becomes empty before the input
+      // node has fully caught up. In this case there will be events where
+      // |(now - mLastEventTime)| is very short, making mLatency negative.
+      // As this happens and the size of |mLatency| becomes greater than
+      // MAX_LATENCY_S, frame dropping starts again to maintain an as short
+      // output queue as possible.
+      float latency = (now - mLastEventTime).ToSeconds();
+      float bufferDuration = aBufferSize / mSampleRate;
+      mLatency += latency - bufferDuration;
+      mLastEventTime = now;
+      if (fabs(mLatency) > MAX_LATENCY_S) {
+        mDroppingBuffers = true;
+      }
+    }
+
+    MutexAutoLock lock(mOutputQueue.Lock());
+    if (mDroppingBuffers) {
+      if (mOutputQueue.ReadyToConsume()) {
+        return;
+      }
+      mDroppingBuffers = false;
+      mLatency = 0;
+    }
+
+    for (uint32_t offset = 0; offset < aBufferSize; offset += WEBAUDIO_BLOCK_SIZE) {
+      AudioChunk& chunk = mOutputQueue.Produce();
+      if (aBuffer) {
+        chunk.mDuration = WEBAUDIO_BLOCK_SIZE;
+        chunk.mBuffer = aBuffer;
+        chunk.mChannelData.SetLength(aBuffer->GetChannels());
+        for (uint32_t i = 0; i < aBuffer->GetChannels(); ++i) {
+          chunk.mChannelData[i] = aBuffer->GetData(i) + offset;
+        }
+        chunk.mVolume = 1.0f;
+        chunk.mBufferFormat = AUDIO_FORMAT_FLOAT32;
+      } else {
+        chunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+
+  // graph thread
+  AudioChunk GetOutputBuffer()
+  {
+    MOZ_ASSERT(!NS_IsMainThread());
+    AudioChunk buffer;
+
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      if (mOutputQueue.ReadyToConsume() > 0) {
+        if (mDelaySoFar == STREAM_TIME_MAX) {
+          mDelaySoFar = 0;
+        }
+        buffer = mOutputQueue.Consume();
+      } else {
+        // If we're out of buffers to consume, just output silence
+        buffer.SetNull(WEBAUDIO_BLOCK_SIZE);
+        if (mDelaySoFar != STREAM_TIME_MAX) {
+          // Remember the delay that we just hit
+          mDelaySoFar += WEBAUDIO_BLOCK_SIZE;
+        }
+      }
+    }
+
+    return buffer;
+  }
+
+  StreamTime DelaySoFar() const
+  {
+    MOZ_ASSERT(!NS_IsMainThread());
+    return mDelaySoFar == STREAM_TIME_MAX ? 0 : mDelaySoFar;
+  }
+
+  void Reset()
+  {
+    MOZ_ASSERT(!NS_IsMainThread());
+    mDelaySoFar = STREAM_TIME_MAX;
+    mLatency = 0.0f;
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      mOutputQueue.Clear();
+    }
+    mLastEventTime = TimeStamp();
+  }
+
+private:
+  OutputQueue mOutputQueue;
+  // How much delay we've seen so far.  This measures the amount of delay
+  // caused by the main thread lagging behind in producing output buffers.
+  // STREAM_TIME_MAX means that we have not received our first buffer yet.
+  StreamTime mDelaySoFar;
+  // The samplerate of the context.
+  float mSampleRate;
+  // This is the latency caused by the buffering. If this grows too high, we
+  // will drop buffers until it is acceptable.
+  float mLatency;
+  // This is the time at which we last produced a buffer, to detect if the main
+  // thread has been blocked.
+  TimeStamp mLastEventTime;
+  // True if we should be dropping buffers.
+  bool mDroppingBuffers;
+};
+
+class ScriptProcessorNodeEngine final : public AudioNodeEngine
+{
+public:
+  ScriptProcessorNodeEngine(ScriptProcessorNode* aNode,
+                            AudioDestinationNode* aDestination,
+                            uint32_t aBufferSize,
+                            uint32_t aNumberOfInputChannels)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    , mSharedBuffers(new SharedBuffers(mDestination->SampleRate()))
+    , mBufferSize(aBufferSize)
+    , mInputChannelCount(aNumberOfInputChannels)
+    , mInputWriteIndex(0)
+  {
+  }
+
+  SharedBuffers* GetSharedBuffers() const
+  {
+    return mSharedBuffers;
+  }
+
+  enum {
+    IS_CONNECTED,
+  };
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
+  {
+    switch (aIndex) {
+      case IS_CONNECTED:
+        mIsConnected = aParam;
+        break;
+      default:
+        NS_ERROR("Bad Int32Parameter");
+    } // End index switch.
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    // This node is not connected to anything. Per spec, we don't fire the
+    // onaudioprocess event. We also want to clear out the input and output
+    // buffer queue, and output a null buffer.
+    if (!mIsConnected) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      mSharedBuffers->Reset();
+      mInputWriteIndex = 0;
+      return;
+    }
+
+    // The input buffer is allocated lazily when non-null input is received.
+    if (!aInput.IsNull() && !mInputBuffer) {
+      mInputBuffer = ThreadSharedFloatArrayBufferList::
+        Create(mInputChannelCount, mBufferSize, fallible);
+      if (mInputBuffer && mInputWriteIndex) {
+        // Zero leading for null chunks that were skipped.
+        for (uint32_t i = 0; i < mInputChannelCount; ++i) {
+          float* channelData = mInputBuffer->GetDataForWrite(i);
+          PodZero(channelData, mInputWriteIndex);
+        }
+      }
+    }
+
+    // First, record our input buffer, if its allocation succeeded.
+    uint32_t inputChannelCount = mInputBuffer ? mInputBuffer->GetChannels() : 0;
+    for (uint32_t i = 0; i < inputChannelCount; ++i) {
+      float* writeData = mInputBuffer->GetDataForWrite(i) + mInputWriteIndex;
+      if (aInput.IsNull()) {
+        PodZero(writeData, aInput.GetDuration());
+      } else {
+        MOZ_ASSERT(aInput.GetDuration() == WEBAUDIO_BLOCK_SIZE, "sanity check");
+        MOZ_ASSERT(aInput.ChannelCount() == inputChannelCount);
+        AudioBlockCopyChannelWithScale(static_cast<const float*>(aInput.mChannelData[i]),
+                                       aInput.mVolume, writeData);
+      }
+    }
+    mInputWriteIndex += aInput.GetDuration();
+
+    // Now, see if we have data to output
+    // Note that we need to do this before sending the buffer to the main
+    // thread so that our delay time is updated.
+    *aOutput = mSharedBuffers->GetOutputBuffer();
+
+    if (mInputWriteIndex >= mBufferSize) {
+      SendBuffersToMainThread(aStream, aFrom);
+      mInputWriteIndex -= mBufferSize;
+    }
+  }
+
+  bool IsActive() const override
+  {
+    // Could return false when !mIsConnected after all output chunks produced
+    // by main thread events calling
+    // SharedBuffers::FinishProducingOutputBuffer() have been processed.
+    return true;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Not owned:
+    // - mDestination (probably)
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mSharedBuffers->SizeOfIncludingThis(aMallocSizeOf);
+    if (mInputBuffer) {
+      amount += mInputBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  void SendBuffersToMainThread(AudioNodeStream* aStream, GraphTime aFrom)
+  {
+    MOZ_ASSERT(!NS_IsMainThread());
+
+    // we now have a full input buffer ready to be sent to the main thread.
+    StreamTime playbackTick = mDestination->GraphTimeToStreamTime(aFrom);
+    // Add the duration of the current sample
+    playbackTick += WEBAUDIO_BLOCK_SIZE;
+    // Add the delay caused by the main thread
+    playbackTick += mSharedBuffers->DelaySoFar();
+    // Compute the playback time in the coordinate system of the destination
+    double playbackTime = mDestination->StreamTimeToSeconds(playbackTick);
+
+    class Command final : public Runnable
+    {
+    public:
+      Command(AudioNodeStream* aStream,
+              already_AddRefed<ThreadSharedFloatArrayBufferList> aInputBuffer,
+              double aPlaybackTime)
+        : mStream(aStream)
+        , mInputBuffer(aInputBuffer)
+        , mPlaybackTime(aPlaybackTime)
+      {
+      }
+
+      NS_IMETHOD Run() override
+      {
+        RefPtr<ThreadSharedFloatArrayBufferList> output;
+
+        auto engine =
+          static_cast<ScriptProcessorNodeEngine*>(mStream->Engine());
+        {
+          auto node = static_cast<ScriptProcessorNode*>
+            (engine->NodeMainThread());
+          if (!node) {
+            return NS_OK;
+          }
+
+          if (node->HasListenersFor(nsGkAtoms::onaudioprocess)) {
+            output = DispatchAudioProcessEvent(node);
+          }
+          // The node may have been destroyed during event dispatch.
+        }
+
+        // Append it to our output buffer queue
+        engine->GetSharedBuffers()->
+          FinishProducingOutputBuffer(output, engine->mBufferSize);
+
+        return NS_OK;
+      }
+
+      // Returns the output buffers if set in event handlers.
+      ThreadSharedFloatArrayBufferList*
+        DispatchAudioProcessEvent(ScriptProcessorNode* aNode)
+      {
+        AudioContext* context = aNode->Context();
+        if (!context) {
+          return nullptr;
+        }
+
+        AutoJSAPI jsapi;
+        if (NS_WARN_IF(!jsapi.Init(aNode->GetOwner()))) {
+          return nullptr;
+        }
+        JSContext* cx = jsapi.cx();
+        uint32_t inputChannelCount = aNode->ChannelCount();
+
+        // Create the input buffer
+        RefPtr<AudioBuffer> inputBuffer;
+        if (mInputBuffer) {
+          ErrorResult rv;
+          inputBuffer =
+            AudioBuffer::Create(context, inputChannelCount,
+                                aNode->BufferSize(), context->SampleRate(),
+                                mInputBuffer.forget(), rv);
+          if (rv.Failed()) {
+            rv.SuppressException();
+            return nullptr;
+          }
+        }
+
+        // Ask content to produce data in the output buffer
+        // Note that we always avoid creating the output buffer here, and we try to
+        // avoid creating the input buffer as well.  The AudioProcessingEvent class
+        // knows how to lazily create them if needed once the script tries to access
+        // them.  Otherwise, we may be able to get away without creating them!
+        RefPtr<AudioProcessingEvent> event =
+          new AudioProcessingEvent(aNode, nullptr, nullptr);
+        event->InitEvent(inputBuffer, inputChannelCount, mPlaybackTime);
+        aNode->DispatchTrustedEvent(event);
+
+        // Steal the output buffers if they have been set.
+        // Don't create a buffer if it hasn't been used to return output;
+        // FinishProducingOutputBuffer() will optimize output = null.
+        // GetThreadSharedChannelsForRate() may also return null after OOM.
+        if (event->HasOutputBuffer()) {
+          ErrorResult rv;
+          AudioBuffer* buffer = event->GetOutputBuffer(rv);
+          // HasOutputBuffer() returning true means that GetOutputBuffer()
+          // will not fail.
+          MOZ_ASSERT(!rv.Failed());
+          return buffer->GetThreadSharedChannelsForRate(cx);
+        }
+
+        return nullptr;
+      }
+    private:
+      RefPtr<AudioNodeStream> mStream;
+      RefPtr<ThreadSharedFloatArrayBufferList> mInputBuffer;
+      double mPlaybackTime;
+    };
+
+    NS_DispatchToMainThread(new Command(aStream, mInputBuffer.forget(),
+                                        playbackTime));
+  }
+
+  friend class ScriptProcessorNode;
+
+  AudioNodeStream* mDestination;
+  nsAutoPtr<SharedBuffers> mSharedBuffers;
+  RefPtr<ThreadSharedFloatArrayBufferList> mInputBuffer;
+  const uint32_t mBufferSize;
+  const uint32_t mInputChannelCount;
+  // The write index into the current input buffer
+  uint32_t mInputWriteIndex;
+  bool mIsConnected = false;
+};
+
+ScriptProcessorNode::ScriptProcessorNode(AudioContext* aContext,
+                                         uint32_t aBufferSize,
+                                         uint32_t aNumberOfInputChannels,
+                                         uint32_t aNumberOfOutputChannels)
+  : AudioNode(aContext,
+              aNumberOfInputChannels,
+              mozilla::dom::ChannelCountMode::Explicit,
+              mozilla::dom::ChannelInterpretation::Speakers)
+  , mBufferSize(aBufferSize ?
+                  aBufferSize : // respect what the web developer requested
+                  4096)         // choose our own buffer size -- 4KB for now
+  , mNumberOfOutputChannels(aNumberOfOutputChannels)
+{
+  MOZ_ASSERT(BufferSize() % WEBAUDIO_BLOCK_SIZE == 0, "Invalid buffer size");
+  ScriptProcessorNodeEngine* engine =
+    new ScriptProcessorNodeEngine(this,
+                                  aContext->Destination(),
+                                  BufferSize(),
+                                  aNumberOfInputChannels);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+ScriptProcessorNode::~ScriptProcessorNode()
+{
+}
+
+size_t
+ScriptProcessorNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+ScriptProcessorNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void
+ScriptProcessorNode::EventListenerAdded(nsIAtom* aType)
+{
+  AudioNode::EventListenerAdded(aType);
+  if (aType == nsGkAtoms::onaudioprocess) {
+    UpdateConnectedStatus();
+  }
+}
+
+void
+ScriptProcessorNode::EventListenerRemoved(nsIAtom* aType)
+{
+  AudioNode::EventListenerRemoved(aType);
+  if (aType == nsGkAtoms::onaudioprocess) {
+    UpdateConnectedStatus();
+  }
+}
+
+JSObject*
+ScriptProcessorNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return ScriptProcessorNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+ScriptProcessorNode::UpdateConnectedStatus()
+{
+  bool isConnected = mHasPhantomInput ||
+    !(OutputNodes().IsEmpty() && OutputParams().IsEmpty()
+      && InputNodes().IsEmpty());
+
+  // Events are queued even when there is no listener because a listener
+  // may be added while events are in the queue.
+  SendInt32ParameterToStream(ScriptProcessorNodeEngine::IS_CONNECTED,
+                             isConnected);
+
+  if (isConnected && HasListenersFor(nsGkAtoms::onaudioprocess)) {
+    MarkActive();
+  } else {
+    MarkInactive();
+  }
+}
+
+} // namespace dom
+} // namespace mozilla
+
diff --git a/dom/media/webaudio/ScriptProcessorNode.h b/dom/media/webaudio/ScriptProcessorNode.h
new file mode 100644
index 0000000000..bd1170e9cb
--- /dev/null
+++ b/dom/media/webaudio/ScriptProcessorNode.h
@@ -0,0 +1,147 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ScriptProcessorNode_h_
+#define ScriptProcessorNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+class SharedBuffers;
+
+class ScriptProcessorNode final : public AudioNode
+{
+public:
+  ScriptProcessorNode(AudioContext* aContext,
+                      uint32_t aBufferSize,
+                      uint32_t aNumberOfInputChannels,
+                      uint32_t aNumberOfOutputChannels);
+
+  NS_DECL_ISUPPORTS_INHERITED
+
+  IMPL_EVENT_HANDLER(audioprocess)
+
+  void EventListenerAdded(nsIAtom* aType) override;
+  void EventListenerRemoved(nsIAtom* aType) override;
+
+  JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioNode* Connect(AudioNode& aDestination, uint32_t aOutput,
+                     uint32_t aInput, ErrorResult& aRv) override
+  {
+    AudioNode* node = AudioNode::Connect(aDestination, aOutput, aInput, aRv);
+    if (!aRv.Failed()) {
+      UpdateConnectedStatus();
+    }
+    return node;
+  }
+
+  void Connect(AudioParam& aDestination, uint32_t aOutput,
+               ErrorResult& aRv) override
+  {
+    AudioNode::Connect(aDestination, aOutput, aRv);
+    if (!aRv.Failed()) {
+      UpdateConnectedStatus();
+    }
+  }
+  void Disconnect(ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(uint32_t aOutput, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void NotifyInputsChanged() override
+  {
+    UpdateConnectedStatus();
+  }
+  void NotifyHasPhantomInput() override
+  {
+    mHasPhantomInput = true;
+    // No need to UpdateConnectedStatus() because there was previously an
+    // input in InputNodes().
+  }
+  void Disconnect(AudioNode& aDestination, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aDestination, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioNode& aDestination, uint32_t aOutput, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aDestination, aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioNode& aDestination, uint32_t aOutput, uint32_t aInput, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aDestination, aOutput, aInput, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioParam& aDestination, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aDestination, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioParam& aDestination, uint32_t aOutput, ErrorResult& aRv) override
+  {
+    AudioNode::Disconnect(aDestination, aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override
+  {
+    if (aChannelCount != ChannelCount()) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    }
+    return;
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode, ErrorResult& aRv) override
+  {
+    if (aMode != ChannelCountMode::Explicit) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    }
+    return;
+  }
+
+  uint32_t BufferSize() const
+  {
+    return mBufferSize;
+  }
+
+  uint32_t NumberOfOutputChannels() const
+  {
+    return mNumberOfOutputChannels;
+  }
+
+  using DOMEventTargetHelper::DispatchTrustedEvent;
+
+  const char* NodeType() const override
+  {
+    return "ScriptProcessorNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+private:
+  virtual ~ScriptProcessorNode();
+
+  void UpdateConnectedStatus();
+
+  const uint32_t mBufferSize;
+  const uint32_t mNumberOfOutputChannels;
+  bool mHasPhantomInput = false;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/StereoPannerNode.cpp b/dom/media/webaudio/StereoPannerNode.cpp
new file mode 100644
index 0000000000..fc804e7b47
--- /dev/null
+++ b/dom/media/webaudio/StereoPannerNode.cpp
@@ -0,0 +1,211 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "StereoPannerNode.h"
+#include "mozilla/dom/StereoPannerNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "AlignmentUtils.h"
+#include "WebAudioUtils.h"
+#include "PanningUtils.h"
+#include "AudioParamTimeline.h"
+#include "AudioParam.h"
+
+namespace mozilla {
+namespace dom {
+
+using namespace std;
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(StereoPannerNode, AudioNode, mPan)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(StereoPannerNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(StereoPannerNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(StereoPannerNode, AudioNode)
+
+class StereoPannerNodeEngine final : public AudioNodeEngine
+{
+public:
+  StereoPannerNodeEngine(AudioNode* aNode,
+                         AudioDestinationNode* aDestination)
+    : AudioNodeEngine(aNode)
+    , mDestination(aDestination->Stream())
+    // Keep the default value in sync with the default value in
+    // StereoPannerNode::StereoPannerNode.
+    , mPan(0.f)
+  {
+  }
+
+  enum Parameters {
+    PAN
+  };
+  void RecvTimelineEvent(uint32_t aIndex,
+                         AudioTimelineEvent& aEvent) override
+  {
+    MOZ_ASSERT(mDestination);
+    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
+                                                    mDestination);
+
+    switch (aIndex) {
+    case PAN:
+      mPan.InsertEvent<int64_t>(aEvent);
+      break;
+    default:
+      NS_ERROR("Bad StereoPannerNode TimelineParameter");
+    }
+  }
+
+  void GetGainValuesForPanning(float aPanning,
+                               bool aMonoToStereo,
+                               float& aLeftGain,
+                               float& aRightGain)
+  {
+    // Clamp and normalize the panning in [0; 1]
+    aPanning = std::min(std::max(aPanning, -1.f), 1.f);
+
+    if (aMonoToStereo) {
+      aPanning += 1;
+      aPanning /= 2;
+    } else if (aPanning <= 0) {
+      aPanning += 1;
+    }
+
+    aLeftGain  = cos(0.5 * M_PI * aPanning);
+    aRightGain = sin(0.5 * M_PI * aPanning);
+  }
+
+  void SetToSilentStereoBlock(AudioBlock* aChunk)
+  {
+    for (uint32_t channel = 0; channel < 2; channel++) {
+      float* samples = aChunk->ChannelFloatsForWrite(channel);
+      for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i++) {
+        samples[i] = 0.f;
+      }
+    }
+  }
+
+  void UpmixToStereoIfNeeded(const AudioBlock& aInput, AudioBlock* aOutput)
+  {
+    if (aInput.ChannelCount() == 2) {
+      const float* inputL = static_cast<const float*>(aInput.mChannelData[0]);
+      const float* inputR = static_cast<const float*>(aInput.mChannelData[1]);
+      float* outputL = aOutput->ChannelFloatsForWrite(0);
+      float* outputR = aOutput->ChannelFloatsForWrite(1);
+
+      AudioBlockCopyChannelWithScale(inputL, aInput.mVolume, outputL);
+      AudioBlockCopyChannelWithScale(inputR, aInput.mVolume, outputR);
+    } else {
+      MOZ_ASSERT(aInput.ChannelCount() == 1);
+      GainMonoToStereo(aInput, aOutput, aInput.mVolume, aInput.mVolume);
+    }
+  }
+
+  virtual void ProcessBlock(AudioNodeStream* aStream,
+                            GraphTime aFrom,
+                            const AudioBlock& aInput,
+                            AudioBlock* aOutput,
+                            bool *aFinished) override
+  {
+    // The output of this node is always stereo, no matter what the inputs are.
+    MOZ_ASSERT(aInput.ChannelCount() <= 2);
+    aOutput->AllocateChannels(2);
+    bool monoToStereo = aInput.ChannelCount() == 1;
+
+    if (aInput.IsNull()) {
+      // If input is silent, so is the output
+      SetToSilentStereoBlock(aOutput);
+    } else if (mPan.HasSimpleValue()) {
+      float panning = mPan.GetValue();
+      // If the panning is 0.0, we can simply copy the input to the
+      // output with gain applied, up-mixing to stereo if needed.
+      if (panning == 0.0f) {
+        UpmixToStereoIfNeeded(aInput, aOutput);
+      } else {
+        // Optimize the case where the panning is constant for this processing
+        // block: we can just apply a constant gain on the left and right
+        // channel
+        float gainL, gainR;
+
+        GetGainValuesForPanning(panning, monoToStereo, gainL, gainR);
+        ApplyStereoPanning(aInput, aOutput,
+                           gainL * aInput.mVolume,
+                           gainR * aInput.mVolume,
+                           panning <= 0);
+      }
+    } else {
+      float computedGain[2*WEBAUDIO_BLOCK_SIZE + 4];
+      bool onLeft[WEBAUDIO_BLOCK_SIZE];
+
+      float values[WEBAUDIO_BLOCK_SIZE];
+      StreamTime tick = mDestination->GraphTimeToStreamTime(aFrom);
+      mPan.GetValuesAtTime(tick, values, WEBAUDIO_BLOCK_SIZE);
+
+      float* alignedComputedGain = ALIGNED16(computedGain);
+      ASSERT_ALIGNED16(alignedComputedGain);
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        float left, right;
+        GetGainValuesForPanning(values[counter], monoToStereo, left, right);
+
+        alignedComputedGain[counter] = left * aInput.mVolume;
+        alignedComputedGain[WEBAUDIO_BLOCK_SIZE + counter] = right * aInput.mVolume;
+        onLeft[counter] = values[counter] <= 0;
+      }
+
+      // Apply the gain to the output buffer
+      ApplyStereoPanning(aInput, aOutput, alignedComputedGain, &alignedComputedGain[WEBAUDIO_BLOCK_SIZE], onLeft);
+    }
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mPan;
+};
+
+StereoPannerNode::StereoPannerNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Clamped_max,
+              ChannelInterpretation::Speakers)
+  , mPan(new AudioParam(this, StereoPannerNodeEngine::PAN, 0.f, "pan"))
+{
+  StereoPannerNodeEngine* engine = new StereoPannerNodeEngine(this, aContext->Destination());
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+StereoPannerNode::~StereoPannerNode()
+{
+}
+
+size_t
+StereoPannerNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mPan->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t
+StereoPannerNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject*
+StereoPannerNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return StereoPannerNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/StereoPannerNode.h b/dom/media/webaudio/StereoPannerNode.h
new file mode 100644
index 0000000000..68204f7571
--- /dev/null
+++ b/dom/media/webaudio/StereoPannerNode.h
@@ -0,0 +1,70 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef StereoPannerNode_h_
+#define StereoPannerNode_h_
+
+#include "AudioNode.h"
+#include "mozilla/dom/StereoPannerNodeBinding.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class StereoPannerNode final : public AudioNode
+{
+public:
+  MOZ_DECLARE_REFCOUNTED_TYPENAME(StereoPannerNode)
+  explicit StereoPannerNode(AudioContext* aContext);
+
+  virtual JSObject* WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  virtual void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override
+  {
+    if (aChannelCount > 2) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  virtual void SetChannelCountModeValue(ChannelCountMode aMode, ErrorResult& aRv) override
+  {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  AudioParam* Pan() const
+  {
+    return mPan;
+  }
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(StereoPannerNode, AudioNode)
+
+  virtual const char* NodeType() const override
+  {
+    return "StereoPannerNode";
+  }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+protected:
+  virtual ~StereoPannerNode();
+
+private:
+  RefPtr<AudioParam> mPan;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/ThreeDPoint.cpp b/dom/media/webaudio/ThreeDPoint.cpp
new file mode 100644
index 0000000000..ad816eb89d
--- /dev/null
+++ b/dom/media/webaudio/ThreeDPoint.cpp
@@ -0,0 +1,49 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/**
+ * Other similar methods can be added if needed.
+ */
+
+#include "ThreeDPoint.h"
+#include "WebAudioUtils.h"
+
+namespace mozilla {
+
+namespace dom {
+
+bool
+ThreeDPoint::FuzzyEqual(const ThreeDPoint& other)
+{
+  return WebAudioUtils::FuzzyEqual(x, other.x) &&
+    WebAudioUtils::FuzzyEqual(y, other.y) &&
+    WebAudioUtils::FuzzyEqual(z, other.z);
+}
+
+ThreeDPoint operator-(const ThreeDPoint& lhs, const ThreeDPoint& rhs)
+{
+  return ThreeDPoint(lhs.x - rhs.x, lhs.y - rhs.y, lhs.z - rhs.z);
+}
+
+ThreeDPoint operator*(const ThreeDPoint& lhs, const ThreeDPoint& rhs)
+{
+  return ThreeDPoint(lhs.x * rhs.x, lhs.y * rhs.y, lhs.z * rhs.z);
+}
+
+ThreeDPoint operator*(const ThreeDPoint& lhs, const double rhs)
+{
+  return ThreeDPoint(lhs.x * rhs, lhs.y * rhs, lhs.z * rhs);
+}
+
+bool operator==(const ThreeDPoint& lhs, const ThreeDPoint& rhs)
+{
+  return lhs.x == rhs.x &&
+         lhs.y == rhs.y &&
+         lhs.z == rhs.z;
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/ThreeDPoint.h b/dom/media/webaudio/ThreeDPoint.h
new file mode 100644
index 0000000000..b6d51e69a3
--- /dev/null
+++ b/dom/media/webaudio/ThreeDPoint.h
@@ -0,0 +1,89 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef ThreeDPoint_h_
+#define ThreeDPoint_h_
+
+#include <cmath>
+#include <algorithm>
+
+namespace mozilla {
+
+namespace dom {
+
+struct ThreeDPoint final
+{
+  ThreeDPoint()
+    : x(0.)
+    , y(0.)
+    , z(0.)
+  {
+  }
+  ThreeDPoint(double aX, double aY, double aZ)
+    : x(aX)
+    , y(aY)
+    , z(aZ)
+  {
+  }
+
+  double Magnitude() const
+  {
+    return sqrt(x * x + y * y + z * z);
+  }
+
+  void Normalize()
+  {
+    // Normalize with the maximum norm first to avoid overflow and underflow.
+    double invMax = 1 / MaxNorm();
+    x *= invMax;
+    y *= invMax;
+    z *= invMax;
+
+    double invDistance = 1 / Magnitude();
+    x *= invDistance;
+    y *= invDistance;
+    z *= invDistance;
+  }
+
+  ThreeDPoint CrossProduct(const ThreeDPoint& rhs) const
+  {
+    return ThreeDPoint(y * rhs.z - z * rhs.y,
+                       z * rhs.x - x * rhs.z,
+                       x * rhs.y - y * rhs.x);
+  }
+
+  double DotProduct(const ThreeDPoint& rhs)
+  {
+    return x * rhs.x + y * rhs.y + z * rhs.z;
+  }
+
+  bool IsZero() const
+  {
+    return x == 0 && y == 0 && z == 0;
+  }
+
+  // For comparing two vectors of close to unit magnitude.
+  bool FuzzyEqual(const ThreeDPoint& other);
+
+  double x, y, z;
+
+private:
+  double MaxNorm() const
+  {
+    return std::max(fabs(x), std::max(fabs(y), fabs(z)));
+  }
+};
+
+ThreeDPoint operator-(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+ThreeDPoint operator*(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+ThreeDPoint operator*(const ThreeDPoint& lhs, const double rhs);
+bool operator==(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/WaveShaperNode.cpp b/dom/media/webaudio/WaveShaperNode.cpp
new file mode 100644
index 0000000000..d5c617dcd4
--- /dev/null
+++ b/dom/media/webaudio/WaveShaperNode.cpp
@@ -0,0 +1,392 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WaveShaperNode.h"
+#include "mozilla/dom/WaveShaperNodeBinding.h"
+#include "AlignmentUtils.h"
+#include "AudioNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "mozilla/PodOperations.h"
+
+namespace mozilla {
+namespace dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(WaveShaperNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(WaveShaperNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+  tmp->ClearCurve();
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(WaveShaperNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE_SCRIPT_OBJECTS
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(WaveShaperNode)
+  NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER
+  NS_IMPL_CYCLE_COLLECTION_TRACE_JS_MEMBER_CALLBACK(mCurve)
+NS_IMPL_CYCLE_COLLECTION_TRACE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(WaveShaperNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(WaveShaperNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(WaveShaperNode, AudioNode)
+
+static uint32_t ValueOf(OverSampleType aType)
+{
+  switch (aType) {
+  case OverSampleType::None: return 1;
+  case OverSampleType::_2x:  return 2;
+  case OverSampleType::_4x:  return 4;
+  default:
+    NS_NOTREACHED("We should never reach here");
+    return 1;
+  }
+}
+
+class Resampler final
+{
+public:
+  Resampler()
+    : mType(OverSampleType::None)
+    , mUpSampler(nullptr)
+    , mDownSampler(nullptr)
+    , mChannels(0)
+    , mSampleRate(0)
+  {
+  }
+
+  ~Resampler()
+  {
+    Destroy();
+  }
+
+  void Reset(uint32_t aChannels, TrackRate aSampleRate, OverSampleType aType)
+  {
+    if (aChannels == mChannels &&
+        aSampleRate == mSampleRate &&
+        aType == mType) {
+      return;
+    }
+
+    mChannels = aChannels;
+    mSampleRate = aSampleRate;
+    mType = aType;
+
+    Destroy();
+
+    if (aType == OverSampleType::None) {
+      mBuffer.Clear();
+      return;
+    }
+
+    mUpSampler = speex_resampler_init(aChannels,
+                                      aSampleRate,
+                                      aSampleRate * ValueOf(aType),
+                                      SPEEX_RESAMPLER_QUALITY_MIN,
+                                      nullptr);
+    mDownSampler = speex_resampler_init(aChannels,
+                                        aSampleRate * ValueOf(aType),
+                                        aSampleRate,
+                                        SPEEX_RESAMPLER_QUALITY_MIN,
+                                        nullptr);
+    mBuffer.SetLength(WEBAUDIO_BLOCK_SIZE*ValueOf(aType));
+  }
+
+  float* UpSample(uint32_t aChannel, const float* aInputData, uint32_t aBlocks)
+  {
+    uint32_t inSamples = WEBAUDIO_BLOCK_SIZE;
+    uint32_t outSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
+    float* outputData = mBuffer.Elements();
+
+    MOZ_ASSERT(mBuffer.Length() == outSamples);
+
+    WebAudioUtils::SpeexResamplerProcess(mUpSampler, aChannel,
+                                         aInputData, &inSamples,
+                                         outputData, &outSamples);
+
+    MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE && outSamples == WEBAUDIO_BLOCK_SIZE*aBlocks);
+
+    return outputData;
+  }
+
+  void DownSample(uint32_t aChannel, float* aOutputData, uint32_t aBlocks)
+  {
+    uint32_t inSamples = WEBAUDIO_BLOCK_SIZE*aBlocks;
+    uint32_t outSamples = WEBAUDIO_BLOCK_SIZE;
+    const float* inputData = mBuffer.Elements();
+
+    MOZ_ASSERT(mBuffer.Length() == inSamples);
+
+    WebAudioUtils::SpeexResamplerProcess(mDownSampler, aChannel,
+                                         inputData, &inSamples,
+                                         aOutputData, &outSamples);
+
+    MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE*aBlocks && outSamples == WEBAUDIO_BLOCK_SIZE);
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+  {
+    size_t amount = 0;
+    // Future: properly measure speex memory
+    amount += aMallocSizeOf(mUpSampler);
+    amount += aMallocSizeOf(mDownSampler);
+    amount += mBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+private:
+  void Destroy()
+  {
+    if (mUpSampler) {
+      speex_resampler_destroy(mUpSampler);
+      mUpSampler = nullptr;
+    }
+    if (mDownSampler) {
+      speex_resampler_destroy(mDownSampler);
+      mDownSampler = nullptr;
+    }
+  }
+
+private:
+  OverSampleType mType;
+  SpeexResamplerState* mUpSampler;
+  SpeexResamplerState* mDownSampler;
+  uint32_t mChannels;
+  TrackRate mSampleRate;
+  nsTArray<float> mBuffer;
+};
+
+class WaveShaperNodeEngine final : public AudioNodeEngine
+{
+public:
+  explicit WaveShaperNodeEngine(AudioNode* aNode)
+    : AudioNodeEngine(aNode)
+    , mType(OverSampleType::None)
+  {
+  }
+
+  enum Parameteres {
+    TYPE
+  };
+
+  void SetRawArrayData(nsTArray<float>& aCurve) override
+  {
+    mCurve.SwapElements(aCurve);
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override
+  {
+    switch (aIndex) {
+    case TYPE:
+      mType = static_cast<OverSampleType>(aValue);
+      break;
+    default:
+      NS_ERROR("Bad WaveShaperNode Int32Parameter");
+    }
+  }
+
+  template <uint32_t blocks>
+  void ProcessCurve(const float* aInputBuffer, float* aOutputBuffer)
+  {
+    for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE*blocks; ++j) {
+      // Index into the curve array based on the amplitude of the
+      // incoming signal by using an amplitude range of [-1, 1] and
+      // performing a linear interpolation of the neighbor values.
+      float index = (mCurve.Length() - 1) * (aInputBuffer[j] + 1.0f) / 2.0f;
+      if (index < 0.0f) {
+        aOutputBuffer[j] = mCurve[0];
+      } else {
+        int32_t indexLower = index;
+        if (static_cast<uint32_t>(indexLower) >= mCurve.Length() - 1) {
+          aOutputBuffer[j] = mCurve[mCurve.Length() - 1];
+        } else {
+          uint32_t indexHigher = indexLower + 1;
+          float interpolationFactor = index - indexLower;
+          aOutputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] +
+                                     interpolationFactor  * mCurve[indexHigher];
+        }
+      }
+    }
+  }
+
+  void ProcessBlock(AudioNodeStream* aStream,
+                    GraphTime aFrom,
+                    const AudioBlock& aInput,
+                    AudioBlock* aOutput,
+                    bool* aFinished) override
+  {
+    uint32_t channelCount = aInput.ChannelCount();
+    if (!mCurve.Length()) {
+      // Optimize the case where we don't have a curve buffer
+      *aOutput = aInput;
+      return;
+    }
+
+    // If the input is null, check to see if non-null output will be produced
+    bool nullInput = false;
+    if (channelCount == 0) {
+      float index = (mCurve.Length() - 1) * 0.5;
+      uint32_t indexLower = index;
+      uint32_t indexHigher = indexLower + 1;
+      float interpolationFactor = index - indexLower;
+      if ((1.0f - interpolationFactor) * mCurve[indexLower] +
+          interpolationFactor * mCurve[indexHigher] == 0.0) {
+        *aOutput = aInput;
+        return;
+      } else {
+        nullInput = true;
+        channelCount = 1;
+      }
+    }
+
+    aOutput->AllocateChannels(channelCount);
+    for (uint32_t i = 0; i < channelCount; ++i) {
+      const float* inputSamples;
+      float scaledInput[WEBAUDIO_BLOCK_SIZE + 4];
+      float* alignedScaledInput = ALIGNED16(scaledInput);
+      ASSERT_ALIGNED16(alignedScaledInput);
+      if (!nullInput) {
+        if (aInput.mVolume != 1.0f) {
+          AudioBlockCopyChannelWithScale(
+              static_cast<const float*>(aInput.mChannelData[i]),
+                                        aInput.mVolume,
+                                        alignedScaledInput);
+          inputSamples = alignedScaledInput;
+        } else {
+          inputSamples = static_cast<const float*>(aInput.mChannelData[i]);
+        }
+      } else {
+        PodZero(alignedScaledInput, WEBAUDIO_BLOCK_SIZE);
+        inputSamples = alignedScaledInput;
+      }
+      float* outputBuffer = aOutput->ChannelFloatsForWrite(i);
+      float* sampleBuffer;
+
+      switch (mType) {
+      case OverSampleType::None:
+        mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::None);
+        ProcessCurve<1>(inputSamples, outputBuffer);
+        break;
+      case OverSampleType::_2x:
+        mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_2x);
+        sampleBuffer = mResampler.UpSample(i, inputSamples, 2);
+        ProcessCurve<2>(sampleBuffer, sampleBuffer);
+        mResampler.DownSample(i, outputBuffer, 2);
+        break;
+      case OverSampleType::_4x:
+        mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_4x);
+        sampleBuffer = mResampler.UpSample(i, inputSamples, 4);
+        ProcessCurve<4>(sampleBuffer, sampleBuffer);
+        mResampler.DownSample(i, outputBuffer, 4);
+        break;
+      default:
+        NS_NOTREACHED("We should never reach here");
+      }
+    }
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mCurve.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    amount += mResampler.SizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+private:
+  nsTArray<float> mCurve;
+  OverSampleType mType;
+  Resampler mResampler;
+};
+
+WaveShaperNode::WaveShaperNode(AudioContext* aContext)
+  : AudioNode(aContext,
+              2,
+              ChannelCountMode::Max,
+              ChannelInterpretation::Speakers)
+  , mCurve(nullptr)
+  , mType(OverSampleType::None)
+{
+  mozilla::HoldJSObjects(this);
+
+  WaveShaperNodeEngine* engine = new WaveShaperNodeEngine(this);
+  mStream = AudioNodeStream::Create(aContext, engine,
+                                    AudioNodeStream::NO_STREAM_FLAGS,
+                                    aContext->Graph());
+}
+
+WaveShaperNode::~WaveShaperNode()
+{
+  ClearCurve();
+}
+
+void
+WaveShaperNode::ClearCurve()
+{
+  mCurve = nullptr;
+  mozilla::DropJSObjects(this);
+}
+
+JSObject*
+WaveShaperNode::WrapObject(JSContext *aCx, JS::Handle<JSObject*> aGivenProto)
+{
+  return WaveShaperNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+void
+WaveShaperNode::SetCurve(const Nullable<Float32Array>& aCurve, ErrorResult& aRv)
+{
+  nsTArray<float> curve;
+  if (!aCurve.IsNull()) {
+    const Float32Array& floats = aCurve.Value();
+
+    floats.ComputeLengthAndData();
+    if (floats.IsShared()) {
+      // Throw if the object is mapping shared memory (must opt in).
+      aRv.ThrowTypeError<MSG_TYPEDARRAY_IS_SHARED>(NS_LITERAL_STRING("Argument of WaveShaperNode.setCurve"));
+      return;
+    }
+
+    uint32_t argLength = floats.Length();
+    if (argLength < 2) {
+      aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+      return;
+    }
+
+    if (!curve.SetLength(argLength, fallible)) {
+      aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+      return;
+    }
+
+    PodCopy(curve.Elements(), floats.Data(), floats.Length());
+
+    mCurve = floats.Obj();
+  } else {
+    mCurve = nullptr;
+  }
+
+  AudioNodeStream* ns = mStream;
+  MOZ_ASSERT(ns, "Why don't we have a stream here?");
+  ns->SetRawArrayData(curve);
+}
+
+void
+WaveShaperNode::SetOversample(OverSampleType aType)
+{
+  mType = aType;
+  SendInt32ParameterToStream(WaveShaperNodeEngine::TYPE, static_cast<int32_t>(aType));
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/WaveShaperNode.h b/dom/media/webaudio/WaveShaperNode.h
new file mode 100644
index 0000000000..b58841ee69
--- /dev/null
+++ b/dom/media/webaudio/WaveShaperNode.h
@@ -0,0 +1,72 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WaveShaperNode_h_
+#define WaveShaperNode_h_
+
+#include "AudioNode.h"
+#include "mozilla/dom/WaveShaperNodeBinding.h"
+#include "mozilla/dom/TypedArray.h"
+
+namespace mozilla {
+namespace dom {
+
+class AudioContext;
+
+class WaveShaperNode final : public AudioNode
+{
+public:
+  explicit WaveShaperNode(AudioContext *aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS_INHERITED(WaveShaperNode, AudioNode)
+
+  JSObject* WrapObject(JSContext *aCx, JS::Handle<JSObject*> aGivenProto) override;
+
+  void GetCurve(JSContext* aCx, JS::MutableHandle<JSObject*> aRetval) const
+  {
+    aRetval.set(mCurve);
+  }
+  void SetCurve(const Nullable<Float32Array>& aData, ErrorResult& aRv);
+
+  OverSampleType Oversample() const
+  {
+    return mType;
+  }
+  void SetOversample(OverSampleType aType);
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    // Possibly track in the future:
+    // - mCurve
+    return AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  const char* NodeType() const override
+  {
+    return "WaveShaperNode";
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
+  {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+protected:
+  virtual ~WaveShaperNode();
+
+private:
+  void ClearCurve();
+
+private:
+  JS::Heap<JSObject*> mCurve;
+  OverSampleType mType;
+};
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/WebAudioUtils.cpp b/dom/media/webaudio/WebAudioUtils.cpp
new file mode 100644
index 0000000000..6289f803b3
--- /dev/null
+++ b/dom/media/webaudio/WebAudioUtils.cpp
@@ -0,0 +1,151 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebAudioUtils.h"
+#include "AudioNodeStream.h"
+#include "blink/HRTFDatabaseLoader.h"
+
+#include "nsContentUtils.h"
+#include "nsIConsoleService.h"
+#include "nsIScriptError.h"
+
+namespace mozilla {
+
+LazyLogModule gWebAudioAPILog("WebAudioAPI");
+
+namespace dom {
+
+void WebAudioUtils::ConvertAudioTimelineEventToTicks(AudioTimelineEvent& aEvent,
+                                                     AudioNodeStream* aDest)
+{
+  aEvent.SetTimeInTicks(
+      aDest->SecondsToNearestStreamTime(aEvent.Time<double>()));
+  aEvent.mTimeConstant *= aDest->SampleRate();
+  aEvent.mDuration *= aDest->SampleRate();
+}
+
+void
+WebAudioUtils::Shutdown()
+{
+  WebCore::HRTFDatabaseLoader::shutdown();
+}
+
+int
+WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                     uint32_t aChannel,
+                                     const float* aIn, uint32_t* aInLen,
+                                     float* aOut, uint32_t* aOutLen)
+{
+#ifdef MOZ_SAMPLE_TYPE_S16
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
+  tmp1.SetLength(*aInLen);
+  tmp2.SetLength(*aOutLen);
+  ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
+  int result = speex_resampler_process_int(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
+  ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
+  return result;
+#else
+  return speex_resampler_process_float(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
+#endif
+}
+
+int
+WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                     uint32_t aChannel,
+                                     const int16_t* aIn, uint32_t* aInLen,
+                                     float* aOut, uint32_t* aOutLen)
+{
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp;
+#ifdef MOZ_SAMPLE_TYPE_S16
+  tmp.SetLength(*aOutLen);
+  int result = speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, tmp.Elements(), aOutLen);
+  ConvertAudioSamples(tmp.Elements(), aOut, *aOutLen);
+  return result;
+#else
+  tmp.SetLength(*aInLen);
+  ConvertAudioSamples(aIn, tmp.Elements(), *aInLen);
+  int result = speex_resampler_process_float(aResampler, aChannel, tmp.Elements(), aInLen, aOut, aOutLen);
+  return result;
+#endif
+}
+
+int
+WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                     uint32_t aChannel,
+                                     const int16_t* aIn, uint32_t* aInLen,
+                                     int16_t* aOut, uint32_t* aOutLen)
+{
+#ifdef MOZ_SAMPLE_TYPE_S16
+  return speex_resampler_process_int(aResampler, aChannel, aIn, aInLen, aOut, aOutLen);
+#else
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp1;
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE*4> tmp2;
+  tmp1.SetLength(*aInLen);
+  tmp2.SetLength(*aOutLen);
+  ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
+  int result = speex_resampler_process_float(aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
+  ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
+  return result;
+#endif
+}
+
+void
+WebAudioUtils::LogToDeveloperConsole(uint64_t aWindowID, const char* aKey)
+{
+  // This implementation is derived from dom/media/VideoUtils.cpp, but we
+  // use a windowID so that the message is delivered to the developer console.
+  // It is similar to ContentUtils::ReportToConsole, but also works off main
+  // thread.
+  if (!NS_IsMainThread()) {
+    nsCOMPtr<nsIRunnable> task =
+      NS_NewRunnableFunction([aWindowID, aKey]() { LogToDeveloperConsole(aWindowID, aKey); });
+    NS_DispatchToMainThread(task.forget(), NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  nsCOMPtr<nsIConsoleService> console(
+    do_GetService("@mozilla.org/consoleservice;1"));
+  if (!console) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  nsAutoCString spec;
+  uint32_t aLineNumber, aColumnNumber;
+  JSContext *cx = nsContentUtils::GetCurrentJSContext();
+  if (cx) {
+    nsJSUtils::GetCallingLocation(cx, spec, &aLineNumber, &aColumnNumber);
+  }
+
+  nsresult rv;
+  nsCOMPtr<nsIScriptError> errorObject =
+      do_CreateInstance(NS_SCRIPTERROR_CONTRACTID, &rv);
+  if (!errorObject) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  nsXPIDLString result;
+  rv = nsContentUtils::GetLocalizedString(nsContentUtils::eDOM_PROPERTIES,
+                                          aKey, result);
+
+  if (NS_FAILED(rv)) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  errorObject->InitWithWindowID(result,
+                                NS_ConvertUTF8toUTF16(spec),
+                                EmptyString(),
+                                aLineNumber, aColumnNumber,
+                                nsIScriptError::warningFlag, "Web Audio",
+                                aWindowID);
+  console->LogMessage(errorObject);
+}
+
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webaudio/WebAudioUtils.h b/dom/media/webaudio/WebAudioUtils.h
new file mode 100644
index 0000000000..c0b27b8379
--- /dev/null
+++ b/dom/media/webaudio/WebAudioUtils.h
@@ -0,0 +1,238 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WebAudioUtils_h_
+#define WebAudioUtils_h_
+
+#include <cmath>
+#include <limits>
+#include "mozilla/TypeTraits.h"
+#include "mozilla/FloatingPoint.h"
+#include "MediaSegment.h"
+
+// Forward declaration
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+namespace mozilla {
+
+class AudioNodeStream;
+
+extern LazyLogModule gWebAudioAPILog;
+#define WEB_AUDIO_API_LOG(...) \
+  MOZ_LOG(gWebAudioAPILog, LogLevel::Debug, (__VA_ARGS__))
+
+namespace dom {
+
+struct AudioTimelineEvent;
+
+namespace WebAudioUtils {
+  // 32 is the minimum required by the spec for createBuffer() and
+  // createScriptProcessor() and matches what is used by Blink.  The limit
+  // protects against large memory allocations.
+  const size_t MaxChannelCount = 32;
+  // AudioContext::CreateBuffer() "must support sample-rates in at least the
+  // range 22050 to 96000."
+  const uint32_t MinSampleRate = 8000;
+  const uint32_t MaxSampleRate = 192000;
+
+  inline bool FuzzyEqual(float v1, float v2)
+  {
+    using namespace std;
+    return fabsf(v1 - v2) < 1e-7f;
+  }
+  inline bool FuzzyEqual(double v1, double v2)
+  {
+    using namespace std;
+    return fabs(v1 - v2) < 1e-7;
+  }
+
+  /**
+   * Computes an exponential smoothing rate for a time based variable
+   * over aDuration seconds.
+   */
+  inline double ComputeSmoothingRate(double aDuration, double aSampleRate)
+  {
+    return 1.0 - std::exp(-1.0 / (aDuration * aSampleRate));
+  }
+
+  /**
+   * Converts an AudioTimelineEvent's floating point time values to tick values
+   * with respect to a destination AudioNodeStream.
+   *
+   * This needs to be called for each AudioTimelineEvent that gets sent to an
+   * AudioNodeEngine, on the engine side where the AudioTimlineEvent is
+   * received.  This means that such engines need to be aware of their
+   * destination streams as well.
+   */
+  void ConvertAudioTimelineEventToTicks(AudioTimelineEvent& aEvent,
+                                        AudioNodeStream* aDest);
+
+  /**
+   * Converts a linear value to decibels.  Returns aMinDecibels if the linear
+   * value is 0.
+   */
+  inline float ConvertLinearToDecibels(float aLinearValue, float aMinDecibels)
+  {
+    return aLinearValue ? 20.0f * std::log10(aLinearValue) : aMinDecibels;
+  }
+
+  /**
+   * Converts a decibel value to a linear value.
+   */
+  inline float ConvertDecibelsToLinear(float aDecibels)
+  {
+    return std::pow(10.0f, 0.05f * aDecibels);
+  }
+
+  /**
+   * Converts a decibel to a linear value.
+   */
+  inline float ConvertDecibelToLinear(float aDecibel)
+  {
+    return std::pow(10.0f, 0.05f * aDecibel);
+  }
+
+  inline void FixNaN(double& aDouble)
+  {
+    if (IsNaN(aDouble) || IsInfinite(aDouble)) {
+      aDouble = 0.0;
+    }
+  }
+
+  inline double DiscreteTimeConstantForSampleRate(double timeConstant, double sampleRate)
+  {
+    return 1.0 - std::exp(-1.0 / (sampleRate * timeConstant));
+  }
+
+  inline bool IsTimeValid(double aTime)
+  {
+    return aTime >= 0 && aTime <= (MEDIA_TIME_MAX >> TRACK_RATE_MAX_BITS);
+  }
+
+  /**
+   * Converts a floating point value to an integral type in a safe and
+   * platform agnostic way.  The following program demonstrates the kinds
+   * of ways things can go wrong depending on the CPU architecture you're
+   * compiling for:
+   *
+   * #include <stdio.h>
+   * volatile float r;
+   * int main()
+   * {
+   *   unsigned int q;
+   *   r = 1e100;
+   *   q = r;
+   *   printf("%f %d\n", r, q);
+   *   r = -1e100;
+   *   q = r;
+   *   printf("%f %d\n", r, q);
+   *   r = 1e15;
+   *   q = r;
+   *   printf("%f %x\n", r, q);
+   *   r = 0/0.;
+   *   q = r;
+   *   printf("%f %d\n", r, q);
+   * }
+   *
+   * This program, when compiled for unsigned int, generates the following
+   * results depending on the architecture:
+   *
+   * x86 and x86-64
+   * ---
+   *  inf 0
+   *  -inf 0
+   *  999999995904.000000 -727384064 d4a50000
+   *  nan 0
+   *
+   * ARM
+   * ---
+   *  inf -1
+   *  -inf 0
+   *  999999995904.000000 -1
+   *  nan 0
+   *
+   * When compiled for int, this program generates the following results:
+   *
+   * x86 and x86-64
+   * ---
+   *  inf -2147483648
+   *  -inf -2147483648
+   *  999999995904.000000 -2147483648
+   *  nan -2147483648
+   *
+   * ARM
+   * ---
+   *  inf 2147483647
+   *  -inf -2147483648
+   *  999999995904.000000 2147483647
+   *  nan 0
+   *
+   * Note that the caller is responsible to make sure that the value
+   * passed to this function is not a NaN.  This function will abort if
+   * it sees a NaN.
+   */
+  template <typename IntType, typename FloatType>
+  IntType TruncateFloatToInt(FloatType f)
+  {
+    using namespace std;
+
+    static_assert(mozilla::IsIntegral<IntType>::value == true,
+                  "IntType must be an integral type");
+    static_assert(mozilla::IsFloatingPoint<FloatType>::value == true,
+                  "FloatType must be a floating point type");
+
+    if (mozilla::IsNaN(f)) {
+      // It is the responsibility of the caller to deal with NaN values.
+      // If we ever get to this point, we have a serious bug to fix.
+      NS_RUNTIMEABORT("We should never see a NaN here");
+    }
+
+    if (f > FloatType(numeric_limits<IntType>::max())) {
+      // If the floating point value is outside of the range of maximum
+      // integral value for this type, just clamp to the maximum value.
+      return numeric_limits<IntType>::max();
+    }
+
+    if (f < FloatType(numeric_limits<IntType>::min())) {
+      // If the floating point value is outside of the range of minimum
+      // integral value for this type, just clamp to the minimum value.
+      return numeric_limits<IntType>::min();
+    }
+
+    // Otherwise, this conversion must be well defined.
+    return IntType(f);
+  }
+
+  void Shutdown();
+
+  int
+  SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                        uint32_t aChannel,
+                        const float* aIn, uint32_t* aInLen,
+                        float* aOut, uint32_t* aOutLen);
+
+  int
+  SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                        uint32_t aChannel,
+                        const int16_t* aIn, uint32_t* aInLen,
+                        float* aOut, uint32_t* aOutLen);
+
+  int
+  SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                        uint32_t aChannel,
+                        const int16_t* aIn, uint32_t* aInLen,
+                        int16_t* aOut, uint32_t* aOutLen);
+
+  void
+  LogToDeveloperConsole(uint64_t aWindowID, const char* aKey);
+
+  } // namespace WebAudioUtils
+
+} // namespace dom
+} // namespace mozilla
+
+#endif
+
diff --git a/dom/media/webaudio/blink/Biquad.cpp b/dom/media/webaudio/blink/Biquad.cpp
new file mode 100644
index 0000000000..3aa5260725
--- /dev/null
+++ b/dom/media/webaudio/blink/Biquad.cpp
@@ -0,0 +1,469 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "Biquad.h"
+
+#include <float.h>
+#include <algorithm>
+#include <math.h>
+
+namespace WebCore {
+
+Biquad::Biquad()
+{
+    // Initialize as pass-thru (straight-wire, no filter effect)
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+
+    reset(); // clear filter memory
+}
+
+Biquad::~Biquad()
+{
+}
+
+void Biquad::process(const float* sourceP, float* destP, size_t framesToProcess)
+{
+    // Create local copies of member variables
+    double x1 = m_x1;
+    double x2 = m_x2;
+    double y1 = m_y1;
+    double y2 = m_y2;
+
+    double b0 = m_b0;
+    double b1 = m_b1;
+    double b2 = m_b2;
+    double a1 = m_a1;
+    double a2 = m_a2;
+
+    for (size_t i = 0; i < framesToProcess; ++i) {
+        // FIXME: this can be optimized by pipelining the multiply adds...
+        double x = sourceP[i];
+        double y = b0*x + b1*x1 + b2*x2 - a1*y1 - a2*y2;
+
+        destP[i] = y;
+
+        // Update state variables
+        x2 = x1;
+        x1 = x;
+        y2 = y1;
+        y1 = y;
+    }
+
+    // Avoid introducing a stream of subnormals when input is silent and the
+    // tail approaches zero.
+    // TODO: Remove this code when Bug 1157635 is fixed.
+    if (x1 == 0.0 && x2 == 0.0 && (y1 != 0.0 || y2 != 0.0) &&
+        fabs(y1) < FLT_MIN && fabs(y2) < FLT_MIN) {
+      // Flush future values to zero (until there is new input).
+      y1 = y2 = 0.0;
+      // Flush calculated values.
+      for (int i = framesToProcess; i-- && fabsf(destP[i]) < FLT_MIN; ) {
+        destP[i] = 0.0f;
+      }
+    }
+    // Local variables back to member.
+    m_x1 = x1;
+    m_x2 = x2;
+    m_y1 = y1;
+    m_y2 = y2;
+}
+
+void Biquad::reset()
+{
+    m_x1 = m_x2 = m_y1 = m_y2 = 0;
+}
+
+void Biquad::setLowpassParams(double cutoff, double resonance)
+{
+    // Limit cutoff to 0 to 1.
+    cutoff = std::max(0.0, std::min(cutoff, 1.0));
+
+    if (cutoff == 1) {
+        // When cutoff is 1, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    } else if (cutoff > 0) {
+        // Compute biquad coefficients for lowpass filter
+        resonance = std::max(0.0, resonance); // can't go negative
+        double g = pow(10.0, -0.05 * resonance);
+        double w0 = M_PI * cutoff;
+        double cos_w0 = cos(w0);
+        double alpha = 0.5 * sin(w0) * g;
+
+        double b1 = 1.0 - cos_w0;
+        double b0 = 0.5 * b1;
+        double b2 = b0;
+        double a0 = 1.0 + alpha;
+        double a1 = -2.0 * cos_w0;
+        double a2 = 1.0 - alpha;
+
+        setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+        // When cutoff is zero, nothing gets through the filter, so set
+        // coefficients up correctly.
+        setNormalizedCoefficients(0, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setHighpassParams(double cutoff, double resonance)
+{
+    // Limit cutoff to 0 to 1.
+    cutoff = std::max(0.0, std::min(cutoff, 1.0));
+
+    if (cutoff == 1) {
+        // The z-transform is 0.
+        setNormalizedCoefficients(0, 0, 0,
+                                  1, 0, 0);
+    } else if (cutoff > 0) {
+        // Compute biquad coefficients for highpass filter
+        resonance = std::max(0.0, resonance); // can't go negative
+        double g = pow(10.0, -0.05 * resonance);
+        double w0 = M_PI * cutoff;
+        double cos_w0 = cos(w0);
+        double alpha = 0.5 * sin(w0) * g;
+
+        double b1 = -1.0 - cos_w0;
+        double b0 = -0.5 * b1;
+        double b2 = b0;
+        double a0 = 1.0 + alpha;
+        double a1 = -2.0 * cos_w0;
+        double a2 = 1.0 - alpha;
+
+        setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+      // When cutoff is zero, we need to be careful because the above
+      // gives a quadratic divided by the same quadratic, with poles
+      // and zeros on the unit circle in the same place. When cutoff
+      // is zero, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setNormalizedCoefficients(double b0, double b1, double b2, double a0, double a1, double a2)
+{
+    double a0Inverse = 1 / a0;
+
+    m_b0 = b0 * a0Inverse;
+    m_b1 = b1 * a0Inverse;
+    m_b2 = b2 * a0Inverse;
+    m_a1 = a1 * a0Inverse;
+    m_a2 = a2 * a0Inverse;
+}
+
+void Biquad::setLowShelfParams(double frequency, double dbGain)
+{
+    // Clip frequencies to between 0 and 1, inclusive.
+    frequency = std::max(0.0, std::min(frequency, 1.0));
+
+    double A = pow(10.0, dbGain / 40);
+
+    if (frequency == 1) {
+        // The z-transform is a constant gain.
+        setNormalizedCoefficients(A * A, 0, 0,
+                                  1, 0, 0);
+    } else if (frequency > 0) {
+        double w0 = M_PI * frequency;
+        double S = 1; // filter slope (1 is max value)
+        double alpha = 0.5 * sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        double k = cos(w0);
+        double k2 = 2 * sqrt(A) * alpha;
+        double aPlusOne = A + 1;
+        double aMinusOne = A - 1;
+
+        double b0 = A * (aPlusOne - aMinusOne * k + k2);
+        double b1 = 2 * A * (aMinusOne - aPlusOne * k);
+        double b2 = A * (aPlusOne - aMinusOne * k - k2);
+        double a0 = aPlusOne + aMinusOne * k + k2;
+        double a1 = -2 * (aMinusOne + aPlusOne * k);
+        double a2 = aPlusOne + aMinusOne * k - k2;
+
+        setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+        // When frequency is 0, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setHighShelfParams(double frequency, double dbGain)
+{
+    // Clip frequencies to between 0 and 1, inclusive.
+    frequency = std::max(0.0, std::min(frequency, 1.0));
+
+    double A = pow(10.0, dbGain / 40);
+
+    if (frequency == 1) {
+        // The z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    } else if (frequency > 0) {
+        double w0 = M_PI * frequency;
+        double S = 1; // filter slope (1 is max value)
+        double alpha = 0.5 * sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        double k = cos(w0);
+        double k2 = 2 * sqrt(A) * alpha;
+        double aPlusOne = A + 1;
+        double aMinusOne = A - 1;
+
+        double b0 = A * (aPlusOne + aMinusOne * k + k2);
+        double b1 = -2 * A * (aMinusOne + aPlusOne * k);
+        double b2 = A * (aPlusOne + aMinusOne * k - k2);
+        double a0 = aPlusOne - aMinusOne * k + k2;
+        double a1 = 2 * (aMinusOne - aPlusOne * k);
+        double a2 = aPlusOne - aMinusOne * k - k2;
+
+        setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+        // When frequency = 0, the filter is just a gain, A^2.
+        setNormalizedCoefficients(A * A, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setPeakingParams(double frequency, double Q, double dbGain)
+{
+    // Clip frequencies to between 0 and 1, inclusive.
+    frequency = std::max(0.0, std::min(frequency, 1.0));
+
+    // Don't let Q go negative, which causes an unstable filter.
+    Q = std::max(0.0, Q);
+
+    double A = pow(10.0, dbGain / 40);
+
+    if (frequency > 0 && frequency < 1) {
+        if (Q > 0) {
+            double w0 = M_PI * frequency;
+            double alpha = sin(w0) / (2 * Q);
+            double k = cos(w0);
+
+            double b0 = 1 + alpha * A;
+            double b1 = -2 * k;
+            double b2 = 1 - alpha * A;
+            double a0 = 1 + alpha / A;
+            double a1 = -2 * k;
+            double a2 = 1 - alpha / A;
+
+            setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When Q = 0, the above formulas have problems. If we look at
+            // the z-transform, we can see that the limit as Q->0 is A^2, so
+            // set the filter that way.
+            setNormalizedCoefficients(A * A, 0, 0,
+                                      1, 0, 0);
+        }
+    } else {
+        // When frequency is 0 or 1, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setAllpassParams(double frequency, double Q)
+{
+    // Clip frequencies to between 0 and 1, inclusive.
+    frequency = std::max(0.0, std::min(frequency, 1.0));
+
+    // Don't let Q go negative, which causes an unstable filter.
+    Q = std::max(0.0, Q);
+
+    if (frequency > 0 && frequency < 1) {
+        if (Q > 0) {
+            double w0 = M_PI * frequency;
+            double alpha = sin(w0) / (2 * Q);
+            double k = cos(w0);
+
+            double b0 = 1 - alpha;
+            double b1 = -2 * k;
+            double b2 = 1 + alpha;
+            double a0 = 1 + alpha;
+            double a1 = -2 * k;
+            double a2 = 1 - alpha;
+
+            setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When Q = 0, the above formulas have problems. If we look at
+            // the z-transform, we can see that the limit as Q->0 is -1, so
+            // set the filter that way.
+            setNormalizedCoefficients(-1, 0, 0,
+                                      1, 0, 0);
+        }
+    } else {
+        // When frequency is 0 or 1, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setNotchParams(double frequency, double Q)
+{
+    // Clip frequencies to between 0 and 1, inclusive.
+    frequency = std::max(0.0, std::min(frequency, 1.0));
+
+    // Don't let Q go negative, which causes an unstable filter.
+    Q = std::max(0.0, Q);
+
+    if (frequency > 0 && frequency < 1) {
+        if (Q > 0) {
+            double w0 = M_PI * frequency;
+            double alpha = sin(w0) / (2 * Q);
+            double k = cos(w0);
+
+            double b0 = 1;
+            double b1 = -2 * k;
+            double b2 = 1;
+            double a0 = 1 + alpha;
+            double a1 = -2 * k;
+            double a2 = 1 - alpha;
+
+            setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When Q = 0, the above formulas have problems. If we look at
+            // the z-transform, we can see that the limit as Q->0 is 0, so
+            // set the filter that way.
+            setNormalizedCoefficients(0, 0, 0,
+                                      1, 0, 0);
+        }
+    } else {
+        // When frequency is 0 or 1, the z-transform is 1.
+        setNormalizedCoefficients(1, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setBandpassParams(double frequency, double Q)
+{
+    // No negative frequencies allowed.
+    frequency = std::max(0.0, frequency);
+
+    // Don't let Q go negative, which causes an unstable filter.
+    Q = std::max(0.0, Q);
+
+    if (frequency > 0 && frequency < 1) {
+        double w0 = M_PI * frequency;
+        if (Q > 0) {
+            double alpha = sin(w0) / (2 * Q);
+            double k = cos(w0);
+
+            double b0 = alpha;
+            double b1 = 0;
+            double b2 = -alpha;
+            double a0 = 1 + alpha;
+            double a1 = -2 * k;
+            double a2 = 1 - alpha;
+
+            setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When Q = 0, the above formulas have problems. If we look at
+            // the z-transform, we can see that the limit as Q->0 is 1, so
+            // set the filter that way.
+            setNormalizedCoefficients(1, 0, 0,
+                                      1, 0, 0);
+        }
+    } else {
+        // When the cutoff is zero, the z-transform approaches 0, if Q
+        // > 0. When both Q and cutoff are zero, the z-transform is
+        // pretty much undefined. What should we do in this case?
+        // For now, just make the filter 0. When the cutoff is 1, the
+        // z-transform also approaches 0.
+        setNormalizedCoefficients(0, 0, 0,
+                                  1, 0, 0);
+    }
+}
+
+void Biquad::setZeroPolePairs(const Complex &zero, const Complex &pole)
+{
+    double b0 = 1;
+    double b1 = -2 * zero.real();
+
+    double zeroMag = abs(zero);
+    double b2 = zeroMag * zeroMag;
+
+    double a1 = -2 * pole.real();
+
+    double poleMag = abs(pole);
+    double a2 = poleMag * poleMag;
+    setNormalizedCoefficients(b0, b1, b2, 1, a1, a2);
+}
+
+void Biquad::setAllpassPole(const Complex &pole)
+{
+    Complex zero = Complex(1, 0) / pole;
+    setZeroPolePairs(zero, pole);
+}
+
+void Biquad::getFrequencyResponse(int nFrequencies,
+                                  const float* frequency,
+                                  float* magResponse,
+                                  float* phaseResponse)
+{
+    // Evaluate the Z-transform of the filter at given normalized
+    // frequency from 0 to 1.  (1 corresponds to the Nyquist
+    // frequency.)
+    //
+    // The z-transform of the filter is
+    //
+    // H(z) = (b0 + b1*z^(-1) + b2*z^(-2))/(1 + a1*z^(-1) + a2*z^(-2))
+    //
+    // Evaluate as
+    //
+    // b0 + (b1 + b2*z1)*z1
+    // --------------------
+    // 1 + (a1 + a2*z1)*z1
+    //
+    // with z1 = 1/z and z = exp(j*pi*frequency). Hence z1 = exp(-j*pi*frequency)
+
+    // Make local copies of the coefficients as a micro-optimization.
+    double b0 = m_b0;
+    double b1 = m_b1;
+    double b2 = m_b2;
+    double a1 = m_a1;
+    double a2 = m_a2;
+
+    for (int k = 0; k < nFrequencies; ++k) {
+        double omega = -M_PI * frequency[k];
+        Complex z = Complex(cos(omega), sin(omega));
+        Complex numerator = b0 + (b1 + b2 * z) * z;
+        Complex denominator = Complex(1, 0) + (a1 + a2 * z) * z;
+        // Strangely enough, using complex division:
+        // e.g. Complex response = numerator / denominator;
+        // fails on our test machines, yielding infinities and NaNs, so we do
+        // things the long way here.
+        double n = norm(denominator);
+        double r = (real(numerator)*real(denominator) + imag(numerator)*imag(denominator)) / n;
+        double i = (imag(numerator)*real(denominator) - real(numerator)*imag(denominator)) / n;
+        std::complex<double> response = std::complex<double>(r, i);
+
+        magResponse[k] = static_cast<float>(abs(response));
+        phaseResponse[k] = static_cast<float>(atan2(imag(response), real(response)));
+    }
+}
+
+} // namespace WebCore
+
diff --git a/dom/media/webaudio/blink/Biquad.h b/dom/media/webaudio/blink/Biquad.h
new file mode 100644
index 0000000000..f266af4414
--- /dev/null
+++ b/dom/media/webaudio/blink/Biquad.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef Biquad_h
+#define Biquad_h
+
+#include <complex>
+
+namespace WebCore {
+
+typedef std::complex<double> Complex;
+
+// A basic biquad (two-zero / two-pole digital filter)
+//
+// It can be configured to a number of common and very useful filters:
+//    lowpass, highpass, shelving, parameteric, notch, allpass, ...
+
+class Biquad {
+public:
+    Biquad();
+    ~Biquad();
+
+    void process(const float* sourceP, float* destP, size_t framesToProcess);
+
+    // frequency is 0 - 1 normalized, resonance and dbGain are in decibels.
+    // Q is a unitless quality factor.
+    void setLowpassParams(double frequency, double resonance);
+    void setHighpassParams(double frequency, double resonance);
+    void setBandpassParams(double frequency, double Q);
+    void setLowShelfParams(double frequency, double dbGain);
+    void setHighShelfParams(double frequency, double dbGain);
+    void setPeakingParams(double frequency, double Q, double dbGain);
+    void setAllpassParams(double frequency, double Q);
+    void setNotchParams(double frequency, double Q);
+
+    // Set the biquad coefficients given a single zero (other zero will be conjugate)
+    // and a single pole (other pole will be conjugate)
+    void setZeroPolePairs(const Complex& zero, const Complex& pole);
+
+    // Set the biquad coefficients given a single pole (other pole will be conjugate)
+    // (The zeroes will be the inverse of the poles)
+    void setAllpassPole(const Complex& pole);
+
+    // Return true iff the next output block will contain sound even with
+    // silent input.
+    bool hasTail() const { return m_y1 || m_y2 || m_x1 || m_x2; }
+
+    // Resets filter state
+    void reset();
+
+    // Filter response at a set of n frequencies. The magnitude and
+    // phase response are returned in magResponse and phaseResponse.
+    // The phase response is in radians.
+    void getFrequencyResponse(int nFrequencies,
+                              const float* frequency,
+                              float* magResponse,
+                              float* phaseResponse);
+private:
+    void setNormalizedCoefficients(double b0, double b1, double b2, double a0, double a1, double a2);
+
+    // Filter coefficients. The filter is defined as
+    //
+    // y[n] + m_a1*y[n-1] + m_a2*y[n-2] = m_b0*x[n] + m_b1*x[n-1] + m_b2*x[n-2].
+    double m_b0;
+    double m_b1;
+    double m_b2;
+    double m_a1;
+    double m_a2;
+
+    // Filter memory
+    //
+    // Double precision for the output values is valuable because errors can
+    // accumulate.  Input values are also stored as double so they need not be
+    // converted again for computation.
+    double m_x1; // input delayed by 1 sample
+    double m_x2; // input delayed by 2 samples
+    double m_y1; // output delayed by 1 sample
+    double m_y2; // output delayed by 2 samples
+};
+
+} // namespace WebCore
+
+#endif // Biquad_h
diff --git a/dom/media/webaudio/blink/DenormalDisabler.h b/dom/media/webaudio/blink/DenormalDisabler.h
new file mode 100644
index 0000000000..241220732d
--- /dev/null
+++ b/dom/media/webaudio/blink/DenormalDisabler.h
@@ -0,0 +1,124 @@
+/*
+ * Copyright (C) 2011, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DenormalDisabler_h
+#define DenormalDisabler_h
+
+#include <cmath>
+#include <float.h>
+
+namespace WebCore {
+
+// Deal with denormals. They can very seriously impact performance on x86.
+
+// Define HAVE_DENORMAL if we support flushing denormals to zero.
+#if defined(XP_WIN) && defined(_MSC_VER)
+#define HAVE_DENORMAL
+#endif
+
+#if defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
+#define HAVE_DENORMAL
+#endif
+
+#ifdef HAVE_DENORMAL
+class DenormalDisabler {
+public:
+    DenormalDisabler()
+            : m_savedCSR(0)
+    {
+#if defined(XP_WIN) && defined(_MSC_VER)
+        // Save the current state, and set mode to flush denormals.
+        //
+        // http://stackoverflow.com/questions/637175/possible-bug-in-controlfp-s-may-not-restore-control-word-correctly
+        _controlfp_s(&m_savedCSR, 0, 0);
+        unsigned int unused;
+        _controlfp_s(&unused, _DN_FLUSH, _MCW_DN);
+#else
+        m_savedCSR = getCSR();
+        setCSR(m_savedCSR | 0x8040);
+#endif
+    }
+
+    ~DenormalDisabler()
+    {
+#if defined(XP_WIN) && defined(_MSC_VER)
+        unsigned int unused;
+        _controlfp_s(&unused, m_savedCSR, _MCW_DN);
+#else
+        setCSR(m_savedCSR);
+#endif
+    }
+
+    // This is a nop if we can flush denormals to zero in hardware.
+    static inline float flushDenormalFloatToZero(float f)
+    {
+#if defined(XP_WIN) && defined(_MSC_VER) && _M_IX86_FP
+        // For systems using x87 instead of sse, there's no hardware support
+        // to flush denormals automatically. Hence, we need to flush
+        // denormals to zero manually.
+        return (fabs(f) < FLT_MIN) ? 0.0f : f;
+#else
+        return f;
+#endif
+    }
+private:
+#if defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
+    inline int getCSR()
+    {
+        int result;
+        asm volatile("stmxcsr %0" : "=m" (result));
+        return result;
+    }
+
+    inline void setCSR(int a)
+    {
+        int temp = a;
+        asm volatile("ldmxcsr %0" : : "m" (temp));
+    }
+
+#endif
+
+    unsigned int m_savedCSR;
+};
+
+#else
+// FIXME: add implementations for other architectures and compilers
+class DenormalDisabler {
+public:
+    DenormalDisabler() { }
+
+    // Assume the worst case that other architectures and compilers
+    // need to flush denormals to zero manually.
+    static inline float flushDenormalFloatToZero(float f)
+    {
+        return (fabs(f) < FLT_MIN) ? 0.0f : f;
+    }
+};
+
+#endif
+
+} // namespace WebCore
+
+#undef HAVE_DENORMAL
+#endif // DenormalDisabler_h
diff --git a/dom/media/webaudio/blink/DynamicsCompressor.cpp b/dom/media/webaudio/blink/DynamicsCompressor.cpp
new file mode 100644
index 0000000000..8f18913c00
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressor.cpp
@@ -0,0 +1,321 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "DynamicsCompressor.h"
+#include "AlignmentUtils.h"
+#include "AudioBlock.h"
+
+#include <cmath>
+#include "AudioNodeEngine.h"
+#include "nsDebug.h"
+
+using mozilla::WEBAUDIO_BLOCK_SIZE;
+using mozilla::AudioBlockCopyChannelWithScale;
+
+namespace WebCore {
+
+DynamicsCompressor::DynamicsCompressor(float sampleRate, unsigned numberOfChannels)
+    : m_numberOfChannels(numberOfChannels)
+    , m_sampleRate(sampleRate)
+    , m_compressor(sampleRate, numberOfChannels)
+{
+    // Uninitialized state - for parameter recalculation.
+    m_lastFilterStageRatio = -1;
+    m_lastAnchor = -1;
+    m_lastFilterStageGain = -1;
+
+    setNumberOfChannels(numberOfChannels);
+    initializeParameters();
+}
+
+size_t DynamicsCompressor::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+    amount += m_preFilterPacks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_preFilterPacks.Length(); i++) {
+        if (m_preFilterPacks[i]) {
+            amount += m_preFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    amount += m_postFilterPacks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_postFilterPacks.Length(); i++) {
+        if (m_postFilterPacks[i]) {
+            amount += m_postFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    amount += aMallocSizeOf(m_sourceChannels.get());
+    amount += aMallocSizeOf(m_destinationChannels.get());
+    amount += m_compressor.sizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+}
+
+void DynamicsCompressor::setParameterValue(unsigned parameterID, float value)
+{
+    MOZ_ASSERT(parameterID < ParamLast);
+    if (parameterID < ParamLast)
+        m_parameters[parameterID] = value;
+}
+
+void DynamicsCompressor::initializeParameters()
+{
+    // Initializes compressor to default values.
+    
+    m_parameters[ParamThreshold] = -24; // dB
+    m_parameters[ParamKnee] = 30; // dB
+    m_parameters[ParamRatio] = 12; // unit-less
+    m_parameters[ParamAttack] = 0.003f; // seconds
+    m_parameters[ParamRelease] = 0.250f; // seconds
+    m_parameters[ParamPreDelay] = 0.006f; // seconds
+
+    // Release zone values 0 -> 1.
+    m_parameters[ParamReleaseZone1] = 0.09f;
+    m_parameters[ParamReleaseZone2] = 0.16f;
+    m_parameters[ParamReleaseZone3] = 0.42f;
+    m_parameters[ParamReleaseZone4] = 0.98f;
+
+    m_parameters[ParamFilterStageGain] = 4.4f; // dB
+    m_parameters[ParamFilterStageRatio] = 2;
+    m_parameters[ParamFilterAnchor] = 15000 / nyquist();
+    
+    m_parameters[ParamPostGain] = 0; // dB
+    m_parameters[ParamReduction] = 0; // dB
+
+    // Linear crossfade (0 -> 1).
+    m_parameters[ParamEffectBlend] = 1;
+}
+
+float DynamicsCompressor::parameterValue(unsigned parameterID)
+{
+    MOZ_ASSERT(parameterID < ParamLast);
+    return m_parameters[parameterID];
+}
+
+void DynamicsCompressor::setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */)
+{
+    float gk = 1 - gain / 20;
+    float f1 = normalizedFrequency * gk;
+    float f2 = normalizedFrequency / gk;
+    float r1 = expf(-f1 * M_PI);
+    float r2 = expf(-f2 * M_PI);
+
+    MOZ_ASSERT(m_numberOfChannels == m_preFilterPacks.Length());
+
+    for (unsigned i = 0; i < m_numberOfChannels; ++i) {
+        // Set pre-filter zero and pole to create an emphasis filter.
+        ZeroPole& preFilter = m_preFilterPacks[i]->filters[stageIndex];
+        preFilter.setZero(r1);
+        preFilter.setPole(r2);
+
+        // Set post-filter with zero and pole reversed to create the de-emphasis filter.
+        // If there were no compressor kernel in between, they would cancel each other out (allpass filter).
+        ZeroPole& postFilter = m_postFilterPacks[i]->filters[stageIndex];
+        postFilter.setZero(r2);
+        postFilter.setPole(r1);
+    }
+}
+
+void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio)
+{
+    setEmphasisStageParameters(0, gain, anchorFreq);
+    setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio);
+    setEmphasisStageParameters(2, gain, anchorFreq / (filterStageRatio * filterStageRatio));
+    setEmphasisStageParameters(3, gain, anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio));
+}
+
+void DynamicsCompressor::process(const AudioBlock* sourceChunk, AudioBlock* destinationChunk, unsigned framesToProcess)
+{
+    // Though numberOfChannels is retrived from destinationBus, we still name it numberOfChannels instead of numberOfDestinationChannels.
+    // It's because we internally match sourceChannels's size to destinationBus by channel up/down mix. Thus we need numberOfChannels
+    // to do the loop work for both m_sourceChannels and m_destinationChannels.
+
+    unsigned numberOfChannels = destinationChunk->ChannelCount();
+    unsigned numberOfSourceChannels = sourceChunk->ChannelCount();
+
+    MOZ_ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);
+
+    if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
+        destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+    }
+
+    switch (numberOfChannels) {
+    case 2: // stereo
+        m_sourceChannels[0] = static_cast<const float*>(sourceChunk->mChannelData[0]);
+
+        if (numberOfSourceChannels > 1)
+            m_sourceChannels[1] = static_cast<const float*>(sourceChunk->mChannelData[1]);
+        else
+            // Simply duplicate mono channel input data to right channel for stereo processing.
+            m_sourceChannels[1] = m_sourceChannels[0];
+
+        break;
+    default:
+        // FIXME : support other number of channels.
+        NS_WARNING("Support other number of channels");
+        destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+    }
+
+    for (unsigned i = 0; i < numberOfChannels; ++i)
+        m_destinationChannels[i] = const_cast<float*>(static_cast<const float*>(
+            destinationChunk->mChannelData[i]));
+
+    float filterStageGain = parameterValue(ParamFilterStageGain);
+    float filterStageRatio = parameterValue(ParamFilterStageRatio);
+    float anchor = parameterValue(ParamFilterAnchor);
+
+    if (filterStageGain != m_lastFilterStageGain || filterStageRatio != m_lastFilterStageRatio || anchor != m_lastAnchor) {
+        m_lastFilterStageGain = filterStageGain;
+        m_lastFilterStageRatio = filterStageRatio;
+        m_lastAnchor = anchor;
+
+        setEmphasisParameters(filterStageGain, anchor, filterStageRatio);
+    }
+
+    float sourceWithVolume[WEBAUDIO_BLOCK_SIZE + 4];
+    float* alignedSourceWithVolume = ALIGNED16(sourceWithVolume);
+    ASSERT_ALIGNED16(alignedSourceWithVolume);
+
+    // Apply pre-emphasis filter.
+    // Note that the final three stages are computed in-place in the destination buffer.
+    for (unsigned i = 0; i < numberOfChannels; ++i) {
+        const float* sourceData;
+        if (sourceChunk->mVolume == 1.0f) {
+          // Fast path, the volume scale doesn't need to get taken into account
+          sourceData = m_sourceChannels[i];
+        } else {
+          AudioBlockCopyChannelWithScale(m_sourceChannels[i],
+                                         sourceChunk->mVolume,
+                                         alignedSourceWithVolume);
+          sourceData = alignedSourceWithVolume;
+        }
+
+        float* destinationData = m_destinationChannels[i];
+        ZeroPole* preFilters = m_preFilterPacks[i]->filters;
+
+        preFilters[0].process(sourceData, destinationData, framesToProcess);
+        preFilters[1].process(destinationData, destinationData, framesToProcess);
+        preFilters[2].process(destinationData, destinationData, framesToProcess);
+        preFilters[3].process(destinationData, destinationData, framesToProcess);
+    }
+
+    float dbThreshold = parameterValue(ParamThreshold);
+    float dbKnee = parameterValue(ParamKnee);
+    float ratio = parameterValue(ParamRatio);
+    float attackTime = parameterValue(ParamAttack);
+    float releaseTime = parameterValue(ParamRelease);
+    float preDelayTime = parameterValue(ParamPreDelay);
+
+    // This is effectively a master volume on the compressed signal (pre-blending).
+    float dbPostGain = parameterValue(ParamPostGain);
+
+    // Linear blending value from dry to completely processed (0 -> 1)
+    // 0 means the signal is completely unprocessed.
+    // 1 mixes in only the compressed signal.
+    float effectBlend = parameterValue(ParamEffectBlend);
+
+    float releaseZone1 = parameterValue(ParamReleaseZone1);
+    float releaseZone2 = parameterValue(ParamReleaseZone2);
+    float releaseZone3 = parameterValue(ParamReleaseZone3);
+    float releaseZone4 = parameterValue(ParamReleaseZone4);
+
+    // Apply compression to the pre-filtered signal.
+    // The processing is performed in place.
+    m_compressor.process(m_destinationChannels.get(),
+                         m_destinationChannels.get(),
+                         numberOfChannels,
+                         framesToProcess,
+
+                         dbThreshold,
+                         dbKnee,
+                         ratio,
+                         attackTime,
+                         releaseTime,
+                         preDelayTime,
+                         dbPostGain,
+                         effectBlend,
+
+                         releaseZone1,
+                         releaseZone2,
+                         releaseZone3,
+                         releaseZone4
+                         );
+                         
+    // Update the compression amount.                     
+    setParameterValue(ParamReduction, m_compressor.meteringGain());
+
+    // Apply de-emphasis filter.
+    for (unsigned i = 0; i < numberOfChannels; ++i) {
+        float* destinationData = m_destinationChannels[i];
+        ZeroPole* postFilters = m_postFilterPacks[i]->filters;
+
+        postFilters[0].process(destinationData, destinationData, framesToProcess);
+        postFilters[1].process(destinationData, destinationData, framesToProcess);
+        postFilters[2].process(destinationData, destinationData, framesToProcess);
+        postFilters[3].process(destinationData, destinationData, framesToProcess);
+    }
+}
+
+void DynamicsCompressor::reset()
+{
+    m_lastFilterStageRatio = -1; // for recalc
+    m_lastAnchor = -1;
+    m_lastFilterStageGain = -1;
+
+    for (unsigned channel = 0; channel < m_numberOfChannels; ++channel) {
+        for (unsigned stageIndex = 0; stageIndex < 4; ++stageIndex) {
+            m_preFilterPacks[channel]->filters[stageIndex].reset();
+            m_postFilterPacks[channel]->filters[stageIndex].reset();
+        }
+    }
+
+    m_compressor.reset();
+}
+
+void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels)
+{
+    if (m_preFilterPacks.Length() == numberOfChannels)
+        return;
+
+    m_preFilterPacks.Clear();
+    m_postFilterPacks.Clear();
+    for (unsigned i = 0; i < numberOfChannels; ++i) {
+        m_preFilterPacks.AppendElement(new ZeroPoleFilterPack4());
+        m_postFilterPacks.AppendElement(new ZeroPoleFilterPack4());
+    }
+
+    m_sourceChannels = mozilla::MakeUnique<const float* []>(numberOfChannels);
+    m_destinationChannels = mozilla::MakeUnique<float* []>(numberOfChannels);
+
+    m_compressor.setNumberOfChannels(numberOfChannels);
+    m_numberOfChannels = numberOfChannels;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/DynamicsCompressor.h b/dom/media/webaudio/blink/DynamicsCompressor.h
new file mode 100644
index 0000000000..f460836b49
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressor.h
@@ -0,0 +1,131 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DynamicsCompressor_h
+#define DynamicsCompressor_h
+
+#include "DynamicsCompressorKernel.h"
+#include "ZeroPole.h"
+
+#include "nsTArray.h"
+#include "nsAutoPtr.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace mozilla {
+class AudioBlock;
+} // namespace mozilla
+
+namespace WebCore {
+
+using mozilla::AudioBlock;
+
+// DynamicsCompressor implements a flexible audio dynamics compression effect such as
+// is commonly used in musical production and game audio. It lowers the volume
+// of the loudest parts of the signal and raises the volume of the softest parts,
+// making the sound richer, fuller, and more controlled.
+
+class DynamicsCompressor {
+public:
+    enum {
+        ParamThreshold,
+        ParamKnee,
+        ParamRatio,
+        ParamAttack,
+        ParamRelease,
+        ParamPreDelay,
+        ParamReleaseZone1,
+        ParamReleaseZone2,
+        ParamReleaseZone3,
+        ParamReleaseZone4,
+        ParamPostGain,
+        ParamFilterStageGain,
+        ParamFilterStageRatio,
+        ParamFilterAnchor,
+        ParamEffectBlend,
+        ParamReduction,
+        ParamLast
+    };
+
+    DynamicsCompressor(float sampleRate, unsigned numberOfChannels);
+
+    void process(const AudioBlock* sourceChunk, AudioBlock* destinationChunk, unsigned framesToProcess);
+    void reset();
+    void setNumberOfChannels(unsigned);
+    unsigned numberOfChannels() const { return m_numberOfChannels; }
+
+    void setParameterValue(unsigned parameterID, float value);
+    float parameterValue(unsigned parameterID);
+
+    float sampleRate() const { return m_sampleRate; }
+    float nyquist() const { return m_sampleRate / 2; }
+
+    double tailTime() const { return 0; }
+    double latencyTime() const { return m_compressor.latencyFrames() / static_cast<double>(sampleRate()); }
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+protected:
+    unsigned m_numberOfChannels;
+
+    // m_parameters holds the tweakable compressor parameters.
+    float m_parameters[ParamLast];
+    void initializeParameters();
+
+    float m_sampleRate;
+
+    // Emphasis filter controls.
+    float m_lastFilterStageRatio;
+    float m_lastAnchor;
+    float m_lastFilterStageGain;
+
+    typedef struct {
+        ZeroPole filters[4];
+        size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+        {
+            return aMallocSizeOf(this);
+        }
+    } ZeroPoleFilterPack4;
+
+    // Per-channel emphasis filters.
+    nsTArray<nsAutoPtr<ZeroPoleFilterPack4> > m_preFilterPacks;
+    nsTArray<nsAutoPtr<ZeroPoleFilterPack4> > m_postFilterPacks;
+
+    mozilla::UniquePtr<const float*[]> m_sourceChannels;
+    mozilla::UniquePtr<float*[]> m_destinationChannels;
+
+    void setEmphasisStageParameters(unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */);
+    void setEmphasisParameters(float gain, float anchorFreq, float filterStageRatio);
+
+    // The core compressor.
+    DynamicsCompressorKernel m_compressor;
+};
+
+} // namespace WebCore
+
+#endif // DynamicsCompressor_h
diff --git a/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp b/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp
new file mode 100644
index 0000000000..e5b4aba2f2
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp
@@ -0,0 +1,491 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "DynamicsCompressorKernel.h"
+
+#include "DenormalDisabler.h"
+#include <algorithm>
+#include <cmath>
+
+#include "mozilla/FloatingPoint.h"
+#include "WebAudioUtils.h"
+
+using namespace std;
+
+using namespace mozilla::dom; // for WebAudioUtils
+using mozilla::IsInfinite;
+using mozilla::IsNaN;
+using mozilla::MakeUnique;
+
+namespace WebCore {
+
+
+// Metering hits peaks instantly, but releases this fast (in seconds).
+const float meteringReleaseTimeConstant = 0.325f;
+
+const float uninitializedValue = -1;
+
+DynamicsCompressorKernel::DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels)
+    : m_sampleRate(sampleRate)
+    , m_lastPreDelayFrames(DefaultPreDelayFrames)
+    , m_preDelayReadIndex(0)
+    , m_preDelayWriteIndex(DefaultPreDelayFrames)
+    , m_ratio(uninitializedValue)
+    , m_slope(uninitializedValue)
+    , m_linearThreshold(uninitializedValue)
+    , m_dbThreshold(uninitializedValue)
+    , m_dbKnee(uninitializedValue)
+    , m_kneeThreshold(uninitializedValue)
+    , m_kneeThresholdDb(uninitializedValue)
+    , m_ykneeThresholdDb(uninitializedValue)
+    , m_K(uninitializedValue)
+{
+    setNumberOfChannels(numberOfChannels);
+
+    // Initializes most member variables
+    reset();
+
+    m_meteringReleaseK =
+        static_cast<float>(WebAudioUtils::DiscreteTimeConstantForSampleRate(meteringReleaseTimeConstant, sampleRate));
+}
+
+size_t DynamicsCompressorKernel::sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = 0;
+    amount += m_preDelayBuffers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_preDelayBuffers.Length(); i++) {
+        amount += aMallocSizeOf(m_preDelayBuffers[i].get());
+    }
+
+    return amount;
+}
+
+void DynamicsCompressorKernel::setNumberOfChannels(unsigned numberOfChannels)
+{
+    if (m_preDelayBuffers.Length() == numberOfChannels)
+        return;
+
+    m_preDelayBuffers.Clear();
+    for (unsigned i = 0; i < numberOfChannels; ++i)
+      m_preDelayBuffers.AppendElement(MakeUnique<float[]>(MaxPreDelayFrames));
+}
+
+void DynamicsCompressorKernel::setPreDelayTime(float preDelayTime)
+{
+    // Re-configure look-ahead section pre-delay if delay time has changed.
+    unsigned preDelayFrames = preDelayTime * sampleRate();
+    if (preDelayFrames > MaxPreDelayFrames - 1)
+        preDelayFrames = MaxPreDelayFrames - 1;
+
+    if (m_lastPreDelayFrames != preDelayFrames) {
+        m_lastPreDelayFrames = preDelayFrames;
+        for (unsigned i = 0; i < m_preDelayBuffers.Length(); ++i)
+	  memset(m_preDelayBuffers[i].get(), 0, sizeof(float) * MaxPreDelayFrames);
+
+        m_preDelayReadIndex = 0;
+        m_preDelayWriteIndex = preDelayFrames;
+    }
+}
+
+// Exponential curve for the knee.
+// It is 1st derivative matched at m_linearThreshold and asymptotically approaches the value m_linearThreshold + 1 / k.
+float DynamicsCompressorKernel::kneeCurve(float x, float k)
+{
+    // Linear up to threshold.
+    if (x < m_linearThreshold)
+        return x;
+
+    return m_linearThreshold + (1 - expf(-k * (x - m_linearThreshold))) / k;
+}
+
+// Full compression curve with constant ratio after knee.
+float DynamicsCompressorKernel::saturate(float x, float k)
+{
+    float y;
+
+    if (x < m_kneeThreshold)
+        y = kneeCurve(x, k);
+    else {
+        // Constant ratio after knee.
+        float xDb = WebAudioUtils::ConvertLinearToDecibels(x, -1000.0f);
+        float yDb = m_ykneeThresholdDb + m_slope * (xDb - m_kneeThresholdDb);
+
+        y = WebAudioUtils::ConvertDecibelsToLinear(yDb);
+    }
+
+    return y;
+}
+
+// Approximate 1st derivative with input and output expressed in dB.
+// This slope is equal to the inverse of the compression "ratio".
+// In other words, a compression ratio of 20 would be a slope of 1/20.
+float DynamicsCompressorKernel::slopeAt(float x, float k)
+{
+    if (x < m_linearThreshold)
+        return 1;
+
+    float x2 = x * 1.001;
+
+    float xDb = WebAudioUtils::ConvertLinearToDecibels(x, -1000.0f);
+    float x2Db = WebAudioUtils::ConvertLinearToDecibels(x2, -1000.0f);
+
+    float yDb = WebAudioUtils::ConvertLinearToDecibels(kneeCurve(x, k), -1000.0f);
+    float y2Db = WebAudioUtils::ConvertLinearToDecibels(kneeCurve(x2, k), -1000.0f);
+
+    float m = (y2Db - yDb) / (x2Db - xDb);
+
+    return m;
+}
+
+float DynamicsCompressorKernel::kAtSlope(float desiredSlope)
+{
+    float xDb = m_dbThreshold + m_dbKnee;
+    float x = WebAudioUtils::ConvertDecibelsToLinear(xDb);
+
+    // Approximate k given initial values.
+    float minK = 0.1f;
+    float maxK = 10000;
+    float k = 5;
+
+    for (int i = 0; i < 15; ++i) {
+        // A high value for k will more quickly asymptotically approach a slope of 0.
+        float slope = slopeAt(x, k);
+
+        if (slope < desiredSlope) {
+            // k is too high.
+            maxK = k;
+        } else {
+            // k is too low.
+            minK = k;
+        }
+
+        // Re-calculate based on geometric mean.
+        k = sqrtf(minK * maxK);
+    }
+
+    return k;
+}
+
+float DynamicsCompressorKernel::updateStaticCurveParameters(float dbThreshold, float dbKnee, float ratio)
+{
+    if (dbThreshold != m_dbThreshold || dbKnee != m_dbKnee || ratio != m_ratio) {
+        // Threshold and knee.
+        m_dbThreshold = dbThreshold;
+        m_linearThreshold = WebAudioUtils::ConvertDecibelsToLinear(dbThreshold);
+        m_dbKnee = dbKnee;
+
+        // Compute knee parameters.
+        m_ratio = ratio;
+        m_slope = 1 / m_ratio;
+
+        float k = kAtSlope(1 / m_ratio);
+
+        m_kneeThresholdDb = dbThreshold + dbKnee;
+        m_kneeThreshold = WebAudioUtils::ConvertDecibelsToLinear(m_kneeThresholdDb);
+
+        m_ykneeThresholdDb = WebAudioUtils::ConvertLinearToDecibels(kneeCurve(m_kneeThreshold, k), -1000.0f);
+
+        m_K = k;
+    }
+    return m_K;
+}
+
+void DynamicsCompressorKernel::process(float* sourceChannels[],
+                                       float* destinationChannels[],
+                                       unsigned numberOfChannels,
+                                       unsigned framesToProcess,
+
+                                       float dbThreshold,
+                                       float dbKnee,
+                                       float ratio,
+                                       float attackTime,
+                                       float releaseTime,
+                                       float preDelayTime,
+                                       float dbPostGain,
+                                       float effectBlend, /* equal power crossfade */
+
+                                       float releaseZone1,
+                                       float releaseZone2,
+                                       float releaseZone3,
+                                       float releaseZone4
+                                       )
+{
+    MOZ_ASSERT(m_preDelayBuffers.Length() == numberOfChannels);
+
+    float sampleRate = this->sampleRate();
+
+    float dryMix = 1 - effectBlend;
+    float wetMix = effectBlend;
+
+    float k = updateStaticCurveParameters(dbThreshold, dbKnee, ratio);
+
+    // Makeup gain.
+    float fullRangeGain = saturate(1, k);
+    float fullRangeMakeupGain = 1 / fullRangeGain;
+
+    // Empirical/perceptual tuning.
+    fullRangeMakeupGain = powf(fullRangeMakeupGain, 0.6f);
+
+    float masterLinearGain = WebAudioUtils::ConvertDecibelsToLinear(dbPostGain) * fullRangeMakeupGain;
+
+    // Attack parameters.
+    attackTime = max(0.001f, attackTime);
+    float attackFrames = attackTime * sampleRate;
+
+    // Release parameters.
+    float releaseFrames = sampleRate * releaseTime;
+
+    // Detector release time.
+    float satReleaseTime = 0.0025f;
+    float satReleaseFrames = satReleaseTime * sampleRate;
+
+    // Create a smooth function which passes through four points.
+
+    // Polynomial of the form
+    // y = a + b*x + c*x^2 + d*x^3 + e*x^4;
+
+    float y1 = releaseFrames * releaseZone1;
+    float y2 = releaseFrames * releaseZone2;
+    float y3 = releaseFrames * releaseZone3;
+    float y4 = releaseFrames * releaseZone4;
+
+    // All of these coefficients were derived for 4th order polynomial curve fitting where the y values
+    // match the evenly spaced x values as follows: (y1 : x == 0, y2 : x == 1, y3 : x == 2, y4 : x == 3)
+    float kA = 0.9999999999999998f*y1 + 1.8432219684323923e-16f*y2 - 1.9373394351676423e-16f*y3 + 8.824516011816245e-18f*y4;
+    float kB = -1.5788320352845888f*y1 + 2.3305837032074286f*y2 - 0.9141194204840429f*y3 + 0.1623677525612032f*y4;
+    float kC = 0.5334142869106424f*y1 - 1.272736789213631f*y2 + 0.9258856042207512f*y3 - 0.18656310191776226f*y4;
+    float kD = 0.08783463138207234f*y1 - 0.1694162967925622f*y2 + 0.08588057951595272f*y3 - 0.00429891410546283f*y4;
+    float kE = -0.042416883008123074f*y1 + 0.1115693827987602f*y2 - 0.09764676325265872f*y3 + 0.028494263462021576f*y4;
+
+    // x ranges from 0 -> 3       0    1    2   3
+    //                           -15  -10  -5   0db
+
+    // y calculates adaptive release frames depending on the amount of compression.
+
+    setPreDelayTime(preDelayTime);
+
+    const int nDivisionFrames = 32;
+
+    const int nDivisions = framesToProcess / nDivisionFrames;
+
+    unsigned frameIndex = 0;
+    for (int i = 0; i < nDivisions; ++i) {
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+        // Calculate desired gain
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+        // Fix gremlins.
+        if (IsNaN(m_detectorAverage))
+            m_detectorAverage = 1;
+        if (IsInfinite(m_detectorAverage))
+            m_detectorAverage = 1;
+
+        float desiredGain = m_detectorAverage;
+
+        // Pre-warp so we get desiredGain after sin() warp below.
+        float scaledDesiredGain = asinf(desiredGain) / (0.5f * M_PI);
+
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+        // Deal with envelopes
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+        // envelopeRate is the rate we slew from current compressor level to the desired level.
+        // The exact rate depends on if we're attacking or releasing and by how much.
+        float envelopeRate;
+
+        bool isReleasing = scaledDesiredGain > m_compressorGain;
+
+        // compressionDiffDb is the difference between current compression level and the desired level.
+        float compressionDiffDb = WebAudioUtils::ConvertLinearToDecibels(m_compressorGain / scaledDesiredGain, -1000.0f);
+
+        if (isReleasing) {
+            // Release mode - compressionDiffDb should be negative dB
+            m_maxAttackCompressionDiffDb = -1;
+
+            // Fix gremlins.
+            if (IsNaN(compressionDiffDb))
+                compressionDiffDb = -1;
+            if (IsInfinite(compressionDiffDb))
+                compressionDiffDb = -1;
+
+            // Adaptive release - higher compression (lower compressionDiffDb)  releases faster.
+
+            // Contain within range: -12 -> 0 then scale to go from 0 -> 3
+            float x = compressionDiffDb;
+            x = max(-12.0f, x);
+            x = min(0.0f, x);
+            x = 0.25f * (x + 12);
+
+            // Compute adaptive release curve using 4th order polynomial.
+            // Normal values for the polynomial coefficients would create a monotonically increasing function.
+            float x2 = x * x;
+            float x3 = x2 * x;
+            float x4 = x2 * x2;
+            float releaseFrames = kA + kB * x + kC * x2 + kD * x3 + kE * x4;
+
+#define kSpacingDb 5
+            float dbPerFrame = kSpacingDb / releaseFrames;
+
+            envelopeRate = WebAudioUtils::ConvertDecibelsToLinear(dbPerFrame);
+        } else {
+            // Attack mode - compressionDiffDb should be positive dB
+
+            // Fix gremlins.
+            if (IsNaN(compressionDiffDb))
+                compressionDiffDb = 1;
+            if (IsInfinite(compressionDiffDb))
+                compressionDiffDb = 1;
+
+            // As long as we're still in attack mode, use a rate based off
+            // the largest compressionDiffDb we've encountered so far.
+            if (m_maxAttackCompressionDiffDb == -1 || m_maxAttackCompressionDiffDb < compressionDiffDb)
+                m_maxAttackCompressionDiffDb = compressionDiffDb;
+
+            float effAttenDiffDb = max(0.5f, m_maxAttackCompressionDiffDb);
+
+            float x = 0.25f / effAttenDiffDb;
+            envelopeRate = 1 - powf(x, 1 / attackFrames);
+        }
+
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+        // Inner loop - calculate shaped power average - apply compression.
+        // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+        {
+            int preDelayReadIndex = m_preDelayReadIndex;
+            int preDelayWriteIndex = m_preDelayWriteIndex;
+            float detectorAverage = m_detectorAverage;
+            float compressorGain = m_compressorGain;
+
+            int loopFrames = nDivisionFrames;
+            while (loopFrames--) {
+                float compressorInput = 0;
+
+                // Predelay signal, computing compression amount from un-delayed version.
+                for (unsigned i = 0; i < numberOfChannels; ++i) {
+                    float* delayBuffer = m_preDelayBuffers[i].get();
+                    float undelayedSource = sourceChannels[i][frameIndex];
+                    delayBuffer[preDelayWriteIndex] = undelayedSource;
+
+                    float absUndelayedSource = undelayedSource > 0 ? undelayedSource : -undelayedSource;
+                    if (compressorInput < absUndelayedSource)
+                        compressorInput = absUndelayedSource;
+                }
+
+                // Calculate shaped power on undelayed input.
+
+                float scaledInput = compressorInput;
+                float absInput = scaledInput > 0 ? scaledInput : -scaledInput;
+
+                // Put through shaping curve.
+                // This is linear up to the threshold, then enters a "knee" portion followed by the "ratio" portion.
+                // The transition from the threshold to the knee is smooth (1st derivative matched).
+                // The transition from the knee to the ratio portion is smooth (1st derivative matched).
+                float shapedInput = saturate(absInput, k);
+
+                float attenuation = absInput <= 0.0001f ? 1 : shapedInput / absInput;
+
+                float attenuationDb = -WebAudioUtils::ConvertLinearToDecibels(attenuation, -1000.0f);
+                attenuationDb = max(2.0f, attenuationDb);
+
+                float dbPerFrame = attenuationDb / satReleaseFrames;
+
+                float satReleaseRate = WebAudioUtils::ConvertDecibelsToLinear(dbPerFrame) - 1;
+
+                bool isRelease = (attenuation > detectorAverage);
+                float rate = isRelease ? satReleaseRate : 1;
+
+                detectorAverage += (attenuation - detectorAverage) * rate;
+                detectorAverage = min(1.0f, detectorAverage);
+
+                // Fix gremlins.
+                if (IsNaN(detectorAverage))
+                    detectorAverage = 1;
+                if (IsInfinite(detectorAverage))
+                    detectorAverage = 1;
+
+                // Exponential approach to desired gain.
+                if (envelopeRate < 1) {
+                    // Attack - reduce gain to desired.
+                    compressorGain += (scaledDesiredGain - compressorGain) * envelopeRate;
+                } else {
+                    // Release - exponentially increase gain to 1.0
+                    compressorGain *= envelopeRate;
+                    compressorGain = min(1.0f, compressorGain);
+                }
+
+                // Warp pre-compression gain to smooth out sharp exponential transition points.
+                float postWarpCompressorGain = sinf(0.5f * M_PI * compressorGain);
+
+                // Calculate total gain using master gain and effect blend.
+                float totalGain = dryMix + wetMix * masterLinearGain * postWarpCompressorGain;
+
+                // Calculate metering.
+                float dbRealGain = 20 * log10(postWarpCompressorGain);
+                if (dbRealGain < m_meteringGain)
+                    m_meteringGain = dbRealGain;
+                else
+                    m_meteringGain += (dbRealGain - m_meteringGain) * m_meteringReleaseK;
+
+                // Apply final gain.
+                for (unsigned i = 0; i < numberOfChannels; ++i) {
+                    float* delayBuffer = m_preDelayBuffers[i].get();
+                    destinationChannels[i][frameIndex] = delayBuffer[preDelayReadIndex] * totalGain;
+                }
+
+                frameIndex++;
+                preDelayReadIndex = (preDelayReadIndex + 1) & MaxPreDelayFramesMask;
+                preDelayWriteIndex = (preDelayWriteIndex + 1) & MaxPreDelayFramesMask;
+            }
+
+            // Locals back to member variables.
+            m_preDelayReadIndex = preDelayReadIndex;
+            m_preDelayWriteIndex = preDelayWriteIndex;
+            m_detectorAverage = DenormalDisabler::flushDenormalFloatToZero(detectorAverage);
+            m_compressorGain = DenormalDisabler::flushDenormalFloatToZero(compressorGain);
+        }
+    }
+}
+
+void DynamicsCompressorKernel::reset()
+{
+    m_detectorAverage = 0;
+    m_compressorGain = 1;
+    m_meteringGain = 1;
+
+    // Predelay section.
+    for (unsigned i = 0; i < m_preDelayBuffers.Length(); ++i)
+        memset(m_preDelayBuffers[i].get(), 0, sizeof(float) * MaxPreDelayFrames);
+
+    m_preDelayReadIndex = 0;
+    m_preDelayWriteIndex = DefaultPreDelayFrames;
+
+    m_maxAttackCompressionDiffDb = -1; // uninitialized state
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/DynamicsCompressorKernel.h b/dom/media/webaudio/blink/DynamicsCompressorKernel.h
new file mode 100644
index 0000000000..39449949c5
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressorKernel.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DynamicsCompressorKernel_h
+#define DynamicsCompressorKernel_h
+
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace WebCore {
+
+class DynamicsCompressorKernel {
+public:
+    DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels);
+
+    void setNumberOfChannels(unsigned);
+
+    // Performs stereo-linked compression.
+    void process(float* sourceChannels[],
+                 float* destinationChannels[],
+                 unsigned numberOfChannels,
+                 unsigned framesToProcess,
+
+                 float dbThreshold,
+                 float dbKnee,
+                 float ratio,
+                 float attackTime,
+                 float releaseTime,
+                 float preDelayTime,
+                 float dbPostGain,
+                 float effectBlend,
+
+                 float releaseZone1,
+                 float releaseZone2,
+                 float releaseZone3,
+                 float releaseZone4
+                 );
+
+    void reset();
+
+    unsigned latencyFrames() const { return m_lastPreDelayFrames; }
+
+    float sampleRate() const { return m_sampleRate; }
+
+    float meteringGain() const { return m_meteringGain; }
+
+    size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+protected:
+    float m_sampleRate;
+
+    float m_detectorAverage;
+    float m_compressorGain;
+
+    // Metering
+    float m_meteringReleaseK;
+    float m_meteringGain;
+
+    // Lookahead section.
+    enum { MaxPreDelayFrames = 1024 };
+    enum { MaxPreDelayFramesMask = MaxPreDelayFrames - 1 };
+    enum { DefaultPreDelayFrames = 256 }; // setPreDelayTime() will override this initial value
+    unsigned m_lastPreDelayFrames;
+    void setPreDelayTime(float);
+
+    nsTArray<mozilla::UniquePtr<float[]>> m_preDelayBuffers;
+    int m_preDelayReadIndex;
+    int m_preDelayWriteIndex;
+
+    float m_maxAttackCompressionDiffDb;
+
+    // Static compression curve.
+    float kneeCurve(float x, float k);
+    float saturate(float x, float k);
+    float slopeAt(float x, float k);
+    float kAtSlope(float desiredSlope);
+
+    float updateStaticCurveParameters(float dbThreshold, float dbKnee, float ratio);
+
+    // Amount of input change in dB required for 1 dB of output change.
+    // This applies to the portion of the curve above m_kneeThresholdDb (see below).
+    float m_ratio;
+    float m_slope; // Inverse ratio.
+
+    // The input to output change below the threshold is linear 1:1.
+    float m_linearThreshold;
+    float m_dbThreshold;
+
+    // m_dbKnee is the number of dB above the threshold before we enter the "ratio" portion of the curve.
+    // m_kneeThresholdDb = m_dbThreshold + m_dbKnee
+    // The portion between m_dbThreshold and m_kneeThresholdDb is the "soft knee" portion of the curve
+    // which transitions smoothly from the linear portion to the ratio portion.
+    float m_dbKnee;
+    float m_kneeThreshold;
+    float m_kneeThresholdDb;
+    float m_ykneeThresholdDb;
+
+    // Internal parameter for the knee portion of the curve.
+    float m_K;
+};
+
+} // namespace WebCore
+
+#endif // DynamicsCompressorKernel_h
diff --git a/dom/media/webaudio/blink/FFTConvolver.cpp b/dom/media/webaudio/blink/FFTConvolver.cpp
new file mode 100644
index 0000000000..8694073ae2
--- /dev/null
+++ b/dom/media/webaudio/blink/FFTConvolver.cpp
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "FFTConvolver.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+FFTConvolver::FFTConvolver(size_t fftSize, size_t renderPhase)
+    : m_frame(fftSize)
+    , m_readWriteIndex(renderPhase % (fftSize / 2))
+{
+    MOZ_ASSERT(fftSize >= 2 * WEBAUDIO_BLOCK_SIZE);
+  m_inputBuffer.SetLength(fftSize);
+  PodZero(m_inputBuffer.Elements(), fftSize);
+  m_outputBuffer.SetLength(fftSize);
+  PodZero(m_outputBuffer.Elements(), fftSize);
+  m_lastOverlapBuffer.SetLength(fftSize / 2);
+  PodZero(m_lastOverlapBuffer.Elements(), fftSize / 2);
+}
+
+size_t FFTConvolver::sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = 0;
+    amount += m_frame.SizeOfExcludingThis(aMallocSizeOf);
+    amount += m_inputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    amount += m_outputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    amount += m_lastOverlapBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+}
+
+size_t FFTConvolver::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+  return aMallocSizeOf(this) + sizeOfExcludingThis(aMallocSizeOf);
+}
+
+const float* FFTConvolver::process(FFTBlock* fftKernel, const float* sourceP)
+{
+    size_t halfSize = fftSize() / 2;
+
+    // WEBAUDIO_BLOCK_SIZE must be an exact multiple of halfSize,
+    // halfSize must be a multiple of WEBAUDIO_BLOCK_SIZE
+    // and > WEBAUDIO_BLOCK_SIZE.
+    MOZ_ASSERT(halfSize % WEBAUDIO_BLOCK_SIZE == 0 &&
+               WEBAUDIO_BLOCK_SIZE <= halfSize);
+
+    // Copy samples to input buffer (note contraint above!)
+    float* inputP = m_inputBuffer.Elements();
+
+    MOZ_ASSERT(sourceP && inputP && m_readWriteIndex + WEBAUDIO_BLOCK_SIZE <= m_inputBuffer.Length());
+
+    memcpy(inputP + m_readWriteIndex, sourceP, sizeof(float) * WEBAUDIO_BLOCK_SIZE);
+
+    float* outputP = m_outputBuffer.Elements();
+    m_readWriteIndex += WEBAUDIO_BLOCK_SIZE;
+
+    // Check if it's time to perform the next FFT
+    if (m_readWriteIndex == halfSize) {
+        // The input buffer is now filled (get frequency-domain version)
+        m_frame.PerformFFT(m_inputBuffer.Elements());
+        m_frame.Multiply(*fftKernel);
+        m_frame.GetInverseWithoutScaling(m_outputBuffer.Elements());
+
+        // Overlap-add 1st half from previous time
+        AudioBufferAddWithScale(m_lastOverlapBuffer.Elements(), 1.0f,
+                                m_outputBuffer.Elements(), halfSize);
+
+        // Finally, save 2nd half of result
+        MOZ_ASSERT(m_outputBuffer.Length() == 2 * halfSize && m_lastOverlapBuffer.Length() == halfSize);
+
+        memcpy(m_lastOverlapBuffer.Elements(), m_outputBuffer.Elements() + halfSize, sizeof(float) * halfSize);
+
+        // Reset index back to start for next time
+        m_readWriteIndex = 0;
+    }
+
+    return outputP + m_readWriteIndex;
+}
+
+void FFTConvolver::reset()
+{
+    PodZero(m_lastOverlapBuffer.Elements(), m_lastOverlapBuffer.Length());
+    m_readWriteIndex = 0;
+}
+
+size_t FFTConvolver::latencyFrames() const
+{
+    return std::max<size_t>(fftSize()/2, WEBAUDIO_BLOCK_SIZE) -
+        WEBAUDIO_BLOCK_SIZE;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/FFTConvolver.h b/dom/media/webaudio/blink/FFTConvolver.h
new file mode 100644
index 0000000000..118c6baef5
--- /dev/null
+++ b/dom/media/webaudio/blink/FFTConvolver.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FFTConvolver_h
+#define FFTConvolver_h
+
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+typedef AlignedTArray<float> AlignedAudioFloatArray;
+using mozilla::FFTBlock;
+
+class FFTConvolver {
+public:
+    // |fftSize| must be a power of two.
+    //
+    // |renderPhase| is the initial offset in the initially zero input buffer.
+    // It is coordinated with the other stages, so they don't all do their
+    // FFTs at the same time.
+    explicit FFTConvolver(size_t fftSize, size_t renderPhase = 0);
+
+    // Process WEBAUDIO_BLOCK_SIZE elements of array |sourceP| and return a
+    // pointer to an output array of the same size.
+    //
+    // |fftKernel| must be pre-scaled for FFTBlock::GetInverseWithoutScaling().
+    //
+    // FIXME: Later, we can do more sophisticated buffering to relax this requirement...
+    const float* process(FFTBlock* fftKernel, const float* sourceP);
+
+    void reset();
+
+    size_t fftSize() const { return m_frame.FFTSize(); }
+
+    // The input to output latency is up to fftSize / 2, but alignment of the
+    // FFTs with the blocks reduces this by one block.
+    size_t latencyFrames() const;
+
+    size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    FFTBlock m_frame;
+
+    // Buffer input until we get fftSize / 2 samples then do an FFT
+    size_t m_readWriteIndex;
+    AlignedAudioFloatArray m_inputBuffer;
+
+    // Stores output which we read a little at a time
+    AlignedAudioFloatArray m_outputBuffer;
+
+    // Saves the 2nd half of the FFT buffer, so we can do an overlap-add with the 1st half of the next one
+    AlignedAudioFloatArray m_lastOverlapBuffer;
+};
+
+} // namespace WebCore
+
+#endif // FFTConvolver_h
diff --git a/dom/media/webaudio/blink/HRTFDatabase.cpp b/dom/media/webaudio/blink/HRTFDatabase.cpp
new file mode 100644
index 0000000000..ef236c8554
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabase.cpp
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFDatabase.h"
+
+#include "HRTFElevation.h"
+
+using namespace std;
+
+namespace WebCore {
+
+const int HRTFDatabase::MinElevation = -45;
+const int HRTFDatabase::MaxElevation = 90;
+const unsigned HRTFDatabase::RawElevationAngleSpacing = 15;
+const unsigned HRTFDatabase::NumberOfRawElevations = 10; // -45 -> +90 (each 15 degrees)
+const unsigned HRTFDatabase::InterpolationFactor = 1;
+const unsigned HRTFDatabase::NumberOfTotalElevations = NumberOfRawElevations * InterpolationFactor;
+
+nsReturnRef<HRTFDatabase> HRTFDatabase::create(float sampleRate)
+{
+    return nsReturnRef<HRTFDatabase>(new HRTFDatabase(sampleRate));
+}
+
+HRTFDatabase::HRTFDatabase(float sampleRate)
+    : m_sampleRate(sampleRate)
+{
+    m_elevations.SetLength(NumberOfTotalElevations);
+
+    unsigned elevationIndex = 0;
+    for (int elevation = MinElevation; elevation <= MaxElevation; elevation += RawElevationAngleSpacing) {
+        nsAutoRef<HRTFElevation> hrtfElevation(HRTFElevation::createBuiltin(elevation, sampleRate));
+        MOZ_ASSERT(hrtfElevation.get());
+        if (!hrtfElevation.get())
+            return;
+        
+        m_elevations[elevationIndex] = hrtfElevation.out();
+        elevationIndex += InterpolationFactor;
+    }
+
+    // Now, go back and interpolate elevations.
+    if (InterpolationFactor > 1) {
+        for (unsigned i = 0; i < NumberOfTotalElevations; i += InterpolationFactor) {
+            unsigned j = (i + InterpolationFactor);
+            if (j >= NumberOfTotalElevations)
+                j = i; // for last elevation interpolate with itself
+
+            // Create the interpolated convolution kernels and delays.
+            for (unsigned jj = 1; jj < InterpolationFactor; ++jj) {
+                float x = static_cast<float>(jj) / static_cast<float>(InterpolationFactor);
+                m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
+                MOZ_ASSERT(m_elevations[i + jj].get());
+            }
+        }
+    }
+}
+
+size_t HRTFDatabase::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+    amount += m_elevations.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_elevations.Length(); i++) {
+      amount += m_elevations[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+}
+
+void HRTFDatabase::getKernelsFromAzimuthElevation(double azimuthBlend, unsigned azimuthIndex, double elevationAngle, HRTFKernel* &kernelL, HRTFKernel* &kernelR,
+                                                  double& frameDelayL, double& frameDelayR)
+{
+    unsigned elevationIndex = indexFromElevationAngle(elevationAngle);
+    MOZ_ASSERT(elevationIndex < m_elevations.Length() && m_elevations.Length() > 0);
+    
+    if (!m_elevations.Length()) {
+        kernelL = 0;
+        kernelR = 0;
+        return;
+    }
+    
+    if (elevationIndex > m_elevations.Length() - 1)
+        elevationIndex = m_elevations.Length() - 1;
+    
+    HRTFElevation* hrtfElevation = m_elevations[elevationIndex].get();
+    MOZ_ASSERT(hrtfElevation);
+    if (!hrtfElevation) {
+        kernelL = 0;
+        kernelR = 0;
+        return;
+    }
+    
+    hrtfElevation->getKernelsFromAzimuth(azimuthBlend, azimuthIndex, kernelL, kernelR, frameDelayL, frameDelayR);
+}                                                     
+
+unsigned HRTFDatabase::indexFromElevationAngle(double elevationAngle)
+{
+    // Clamp to allowed range.
+    elevationAngle = mozilla::clamped(elevationAngle,
+                                      static_cast<double>(MinElevation),
+                                      static_cast<double>(MaxElevation));
+
+    unsigned elevationIndex = static_cast<int>(InterpolationFactor * (elevationAngle - MinElevation) / RawElevationAngleSpacing);    
+    return elevationIndex;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFDatabase.h b/dom/media/webaudio/blink/HRTFDatabase.h
new file mode 100644
index 0000000000..400763b8f3
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabase.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFDatabase_h
+#define HRTFDatabase_h
+
+#include "HRTFElevation.h"
+#include "nsAutoRef.h"
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+class HRTFKernel;
+
+class HRTFDatabase {
+public:
+    static nsReturnRef<HRTFDatabase> create(float sampleRate);
+
+    // getKernelsFromAzimuthElevation() returns a left and right ear kernel, and an interpolated left and right frame delay for the given azimuth and elevation.
+    // azimuthBlend must be in the range 0 -> 1.
+    // Valid values for azimuthIndex are 0 -> HRTFElevation::NumberOfTotalAzimuths - 1 (corresponding to angles of 0 -> 360).
+    // Valid values for elevationAngle are MinElevation -> MaxElevation.
+    void getKernelsFromAzimuthElevation(double azimuthBlend, unsigned azimuthIndex, double elevationAngle, HRTFKernel* &kernelL, HRTFKernel* &kernelR, double& frameDelayL, double& frameDelayR);
+
+    // Returns the number of different azimuth angles.
+    static unsigned numberOfAzimuths() { return HRTFElevation::NumberOfTotalAzimuths; }
+
+    float sampleRate() const { return m_sampleRate; }
+
+    // Number of elevations loaded from resource.
+    static const unsigned NumberOfRawElevations;
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    HRTFDatabase(const HRTFDatabase& other) = delete;
+    void operator=(const HRTFDatabase& other) = delete;
+
+    explicit HRTFDatabase(float sampleRate);
+
+    // Minimum and maximum elevation angles (inclusive) for a HRTFDatabase.
+    static const int MinElevation;
+    static const int MaxElevation;
+    static const unsigned RawElevationAngleSpacing;
+
+    // Interpolates by this factor to get the total number of elevations from every elevation loaded from resource.
+    static const unsigned InterpolationFactor;
+    
+    // Total number of elevations after interpolation.
+    static const unsigned NumberOfTotalElevations;
+
+    // Returns the index for the correct HRTFElevation given the elevation angle.
+    static unsigned indexFromElevationAngle(double);
+
+    nsTArray<nsAutoRef<HRTFElevation> > m_elevations;
+    float m_sampleRate;
+};
+
+} // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFDatabase> :
+    public nsPointerRefTraits<WebCore::HRTFDatabase> {
+public:
+    static void Release(WebCore::HRTFDatabase* ptr) { delete(ptr); }
+};
+
+#endif // HRTFDatabase_h
diff --git a/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp b/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp
new file mode 100644
index 0000000000..090e1b2172
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp
@@ -0,0 +1,223 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFDatabaseLoader.h"
+#include "HRTFDatabase.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+// Singleton
+nsTHashtable<HRTFDatabaseLoader::LoaderByRateEntry>*
+    HRTFDatabaseLoader::s_loaderMap = nullptr;
+
+size_t HRTFDatabaseLoader::sizeOfLoaders(mozilla::MallocSizeOf aMallocSizeOf)
+{
+    return s_loaderMap ? s_loaderMap->SizeOfIncludingThis(aMallocSizeOf) : 0;
+}
+
+already_AddRefed<HRTFDatabaseLoader> HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate)
+{
+    MOZ_ASSERT(NS_IsMainThread());
+
+    RefPtr<HRTFDatabaseLoader> loader;
+    
+    if (!s_loaderMap) {
+        s_loaderMap = new nsTHashtable<LoaderByRateEntry>();
+    }
+
+    LoaderByRateEntry* entry = s_loaderMap->PutEntry(sampleRate);
+    loader = entry->mLoader;
+    if (loader) { // existing entry
+        MOZ_ASSERT(sampleRate == loader->databaseSampleRate());
+        return loader.forget();
+    }
+
+    loader = new HRTFDatabaseLoader(sampleRate);
+    entry->mLoader = loader;
+
+    loader->loadAsynchronously();
+
+    return loader.forget();
+}
+
+HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate)
+    : m_refCnt(0)
+    , m_threadLock("HRTFDatabaseLoader")
+    , m_databaseLoaderThread(nullptr)
+    , m_databaseSampleRate(sampleRate)
+{
+    MOZ_ASSERT(NS_IsMainThread());
+}
+
+HRTFDatabaseLoader::~HRTFDatabaseLoader()
+{
+    MOZ_ASSERT(NS_IsMainThread());
+
+    waitForLoaderThreadCompletion();
+    m_hrtfDatabase.reset();
+
+    if (s_loaderMap) {
+        // Remove ourself from the map.
+        s_loaderMap->RemoveEntry(m_databaseSampleRate);
+        if (s_loaderMap->Count() == 0) {
+            delete s_loaderMap;
+            s_loaderMap = nullptr;
+        }
+    }
+}
+
+size_t HRTFDatabaseLoader::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    // NB: Need to make sure we're not competing with the loader thread.
+    const_cast<HRTFDatabaseLoader*>(this)->waitForLoaderThreadCompletion();
+
+    if (m_hrtfDatabase) {
+        amount += m_hrtfDatabase->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+}
+
+class HRTFDatabaseLoader::ProxyReleaseEvent final : public Runnable {
+public:
+    explicit ProxyReleaseEvent(HRTFDatabaseLoader* loader) : mLoader(loader) {}
+    NS_IMETHOD Run() override
+    {
+        mLoader->MainThreadRelease();
+        return NS_OK;
+    }
+private:
+    HRTFDatabaseLoader* mLoader;
+};
+
+void HRTFDatabaseLoader::ProxyRelease()
+{
+    nsCOMPtr<nsIThread> mainThread = do_GetMainThread();
+    if (MOZ_LIKELY(mainThread)) {
+        RefPtr<ProxyReleaseEvent> event = new ProxyReleaseEvent(this);
+        DebugOnly<nsresult> rv =
+            mainThread->Dispatch(event, NS_DISPATCH_NORMAL);
+        MOZ_ASSERT(NS_SUCCEEDED(rv), "Failed to dispatch release event");
+    } else {
+        // Should be in XPCOM shutdown.
+        MOZ_ASSERT(NS_IsMainThread(),
+                   "Main thread is not available for dispatch.");
+        MainThreadRelease();
+    }
+}
+
+void HRTFDatabaseLoader::MainThreadRelease()
+{
+    MOZ_ASSERT(NS_IsMainThread());
+    int count = --m_refCnt;
+    MOZ_ASSERT(count >= 0, "extra release");
+    NS_LOG_RELEASE(this, count, "HRTFDatabaseLoader");
+    if (count == 0) {
+        // It is safe to delete here as the first reference can only be added
+        // on this (main) thread.
+        delete this;
+    }
+}
+
+// Asynchronously load the database in this thread.
+static void databaseLoaderEntry(void* threadData)
+{
+    PR_SetCurrentThreadName("HRTFDatabaseLdr");
+
+    HRTFDatabaseLoader* loader = reinterpret_cast<HRTFDatabaseLoader*>(threadData);
+    MOZ_ASSERT(loader);
+    loader->load();
+}
+
+void HRTFDatabaseLoader::load()
+{
+    MOZ_ASSERT(!NS_IsMainThread());
+    MOZ_ASSERT(!m_hrtfDatabase.get(), "Called twice");
+    // Load the default HRTF database.
+    m_hrtfDatabase = HRTFDatabase::create(m_databaseSampleRate);
+    // Notifies the main thread of completion.  See loadAsynchronously().
+    Release();
+}
+
+void HRTFDatabaseLoader::loadAsynchronously()
+{
+    MOZ_ASSERT(NS_IsMainThread());
+    MOZ_ASSERT(m_refCnt, "Must not be called before a reference is added");
+
+    // Add a reference so that the destructor won't run and wait for the
+    // loader thread, until load() has completed.
+    AddRef();
+
+    MutexAutoLock locker(m_threadLock);
+    
+    MOZ_ASSERT(!m_hrtfDatabase.get() && !m_databaseLoaderThread,
+               "Called twice");
+    // Start the asynchronous database loading process.
+    m_databaseLoaderThread =
+        PR_CreateThread(PR_USER_THREAD, databaseLoaderEntry, this,
+                        PR_PRIORITY_NORMAL, PR_GLOBAL_THREAD,
+                        PR_JOINABLE_THREAD, 0);
+}
+
+bool HRTFDatabaseLoader::isLoaded() const
+{
+    return m_hrtfDatabase.get();
+}
+
+void HRTFDatabaseLoader::waitForLoaderThreadCompletion()
+{
+    MutexAutoLock locker(m_threadLock);
+    
+    // waitForThreadCompletion() should not be called twice for the same thread.
+    if (m_databaseLoaderThread) {
+        DebugOnly<PRStatus> status = PR_JoinThread(m_databaseLoaderThread);
+        MOZ_ASSERT(status == PR_SUCCESS, "PR_JoinThread failed");
+    }
+    m_databaseLoaderThread = nullptr;
+}
+
+void HRTFDatabaseLoader::shutdown()
+{
+    MOZ_ASSERT(NS_IsMainThread());
+    if (s_loaderMap) {
+        // Set s_loaderMap to nullptr so that the hashtable is not modified on
+        // reference release during enumeration.
+        nsTHashtable<LoaderByRateEntry>* loaderMap = s_loaderMap;
+        s_loaderMap = nullptr;
+        for (auto iter = loaderMap->Iter(); !iter.Done(); iter.Next()) {
+          iter.Get()->mLoader->waitForLoaderThreadCompletion();
+        }
+        delete loaderMap;
+    }
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFDatabaseLoader.h b/dom/media/webaudio/blink/HRTFDatabaseLoader.h
new file mode 100644
index 0000000000..50a875b183
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabaseLoader.h
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFDatabaseLoader_h
+#define HRTFDatabaseLoader_h
+
+#include "nsHashKeys.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/Mutex.h"
+#include "HRTFDatabase.h"
+
+template <class EntryType> class nsTHashtable;
+template <class T> class nsAutoRef;
+
+namespace WebCore {
+
+// HRTFDatabaseLoader will asynchronously load the default HRTFDatabase in a new thread.
+
+class HRTFDatabaseLoader {
+public:
+    // Lazily creates a HRTFDatabaseLoader (if not already created) for the given sample-rate
+    // and starts loading asynchronously (when created the first time).
+    // Returns the HRTFDatabaseLoader.
+    // Must be called from the main thread.
+    static already_AddRefed<HRTFDatabaseLoader> createAndLoadAsynchronouslyIfNecessary(float sampleRate);
+
+    // AddRef and Release may be called from any thread.
+    void AddRef()
+    {
+#if defined(DEBUG) || defined(NS_BUILD_REFCNT_LOGGING)
+        int count =
+#endif
+          ++m_refCnt;
+        MOZ_ASSERT(count > 0, "invalid ref count");
+        NS_LOG_ADDREF(this, count, "HRTFDatabaseLoader", sizeof(*this));
+    }
+
+    void Release()
+    {
+        // The last reference can't be removed on a non-main thread because
+        // the object can be accessed on the main thread from the hash
+        // table via createAndLoadAsynchronouslyIfNecessary().
+        int count = m_refCnt;
+        MOZ_ASSERT(count > 0, "extra release");
+        // Optimization attempt to possibly skip proxying the release to the
+        // main thread.
+        if (count != 1 && m_refCnt.compareExchange(count, count - 1)) {
+            NS_LOG_RELEASE(this, count - 1, "HRTFDatabaseLoader");
+            return;
+        }
+
+        ProxyRelease();
+    }
+
+    // Returns true once the default database has been completely loaded.
+    bool isLoaded() const;
+
+    // waitForLoaderThreadCompletion() may be called more than once,
+    // on any thread except m_databaseLoaderThread.
+    void waitForLoaderThreadCompletion();
+    
+    HRTFDatabase* database() { return m_hrtfDatabase.get(); }
+
+    float databaseSampleRate() const { return m_databaseSampleRate; }
+
+    static void shutdown();
+    
+    // Called in asynchronous loading thread.
+    void load();
+
+    // Sums the size of all cached database loaders.
+    static size_t sizeOfLoaders(mozilla::MallocSizeOf aMallocSizeOf);
+
+private:
+    // Both constructor and destructor must be called from the main thread.
+    explicit HRTFDatabaseLoader(float sampleRate);
+    ~HRTFDatabaseLoader();
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+    void ProxyRelease(); // any thread
+    void MainThreadRelease(); // main thread only
+    class ProxyReleaseEvent;
+
+    // If it hasn't already been loaded, creates a new thread and initiates asynchronous loading of the default database.
+    // This must be called from the main thread.
+    void loadAsynchronously();
+
+    // Map from sample-rate to loader.
+    class LoaderByRateEntry : public nsFloatHashKey {
+    public:
+        explicit LoaderByRateEntry(KeyTypePointer aKey)
+            : nsFloatHashKey(aKey)
+            , mLoader() // so PutEntry() will zero-initialize
+        {
+        }
+
+        size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+        {
+            return mLoader ? mLoader->sizeOfIncludingThis(aMallocSizeOf) : 0;
+        }
+
+        HRTFDatabaseLoader* mLoader;
+    };
+
+    // Keeps track of loaders on a per-sample-rate basis.
+    static nsTHashtable<LoaderByRateEntry> *s_loaderMap; // singleton
+
+    mozilla::Atomic<int> m_refCnt;
+
+    nsAutoRef<HRTFDatabase> m_hrtfDatabase;
+
+    // Holding a m_threadLock is required when accessing m_databaseLoaderThread.
+    mozilla::Mutex m_threadLock;
+    PRThread* m_databaseLoaderThread;
+
+    float m_databaseSampleRate;
+};
+
+} // namespace WebCore
+
+#endif // HRTFDatabaseLoader_h
diff --git a/dom/media/webaudio/blink/HRTFElevation.cpp b/dom/media/webaudio/blink/HRTFElevation.cpp
new file mode 100644
index 0000000000..2300872f3a
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFElevation.cpp
@@ -0,0 +1,328 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFElevation.h"
+
+#include <speex/speex_resampler.h>
+#include "mozilla/PodOperations.h"
+#include "AudioSampleFormat.h"
+
+#include "IRC_Composite_C_R0195-incl.cpp"
+
+using namespace std;
+using namespace mozilla;
+ 
+namespace WebCore {
+
+const int elevationSpacing = irc_composite_c_r0195_elevation_interval;
+const int firstElevation = irc_composite_c_r0195_first_elevation;
+const int numberOfElevations = MOZ_ARRAY_LENGTH(irc_composite_c_r0195);
+
+const unsigned HRTFElevation::NumberOfTotalAzimuths = 360 / 15 * 8;
+
+const int rawSampleRate = irc_composite_c_r0195_sample_rate;
+
+// Number of frames in an individual impulse response.
+const size_t ResponseFrameSize = 256;
+
+size_t HRTFElevation::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    amount += m_kernelListL.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_kernelListL.Length(); i++) {
+        amount += m_kernelListL[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+}
+
+size_t HRTFElevation::fftSizeForSampleRate(float sampleRate)
+{
+    // The IRCAM HRTF impulse responses were 512 sample-frames @44.1KHz,
+    // but these have been truncated to 256 samples.
+    // An FFT-size of twice impulse response size is used (for convolution).
+    // So for sample rates of 44.1KHz an FFT size of 512 is good.
+    // We double the FFT-size only for sample rates at least double this.
+    // If the FFT size is too large then the impulse response will be padded
+    // with zeros without the fade-out provided by HRTFKernel.
+    MOZ_ASSERT(sampleRate > 1.0 && sampleRate < 1048576.0);
+
+    // This is the size if we were to use all raw response samples.
+    unsigned resampledLength =
+        floorf(ResponseFrameSize * sampleRate / rawSampleRate);
+    // Keep things semi-sane, with max FFT size of 1024.
+    unsigned size = min(resampledLength, 1023U);
+    // Ensure a minimum of 2 * WEBAUDIO_BLOCK_SIZE (with the size++ below) for
+    // FFTConvolver and set the 8 least significant bits for rounding up to
+    // the next power of 2 below.
+    size |= 2 * WEBAUDIO_BLOCK_SIZE - 1;
+    // Round up to the next power of 2, making the FFT size no more than twice
+    // the impulse response length.  This doubles size for values that are
+    // already powers of 2.  This works by filling in alls bit to right of the
+    // most significant bit.  The most significant bit is no greater than
+    // 1 << 9, and the least significant 8 bits were already set above, so
+    // there is at most one bit to add.
+    size |= (size >> 1);
+    size++;
+    MOZ_ASSERT((size & (size - 1)) == 0);
+
+    return size;
+}
+
+nsReturnRef<HRTFKernel> HRTFElevation::calculateKernelForAzimuthElevation(int azimuth, int elevation, SpeexResamplerState* resampler, float sampleRate)
+{
+    int elevationIndex = (elevation - firstElevation) / elevationSpacing;
+    MOZ_ASSERT(elevationIndex >= 0 && elevationIndex <= numberOfElevations);
+
+    int numberOfAzimuths = irc_composite_c_r0195[elevationIndex].count;
+    int azimuthSpacing = 360 / numberOfAzimuths;
+    MOZ_ASSERT(numberOfAzimuths * azimuthSpacing == 360);
+
+    int azimuthIndex = azimuth / azimuthSpacing;
+    MOZ_ASSERT(azimuthIndex * azimuthSpacing == azimuth);
+
+    const int16_t (&impulse_response_data)[ResponseFrameSize] =
+        irc_composite_c_r0195[elevationIndex].azimuths[azimuthIndex];
+
+    // When libspeex_resampler is compiled with FIXED_POINT, samples in
+    // speex_resampler_process_float are rounded directly to int16_t, which
+    // only works well if the floats are in the range +/-32767.  On such
+    // platforms it's better to resample before converting to float anyway.
+#ifdef MOZ_SAMPLE_TYPE_S16
+#  define RESAMPLER_PROCESS speex_resampler_process_int
+    const int16_t* response = impulse_response_data;
+    const int16_t* resampledResponse;
+#else
+#  define RESAMPLER_PROCESS speex_resampler_process_float
+    float response[ResponseFrameSize];
+    ConvertAudioSamples(impulse_response_data, response, ResponseFrameSize);
+    float* resampledResponse;
+#endif
+
+    // Note that depending on the fftSize returned by the panner, we may be truncating the impulse response.
+    const size_t resampledResponseLength = fftSizeForSampleRate(sampleRate) / 2;
+
+    AutoTArray<AudioDataValue, 2 * ResponseFrameSize> resampled;
+    if (sampleRate == rawSampleRate) {
+        resampledResponse = response;
+        MOZ_ASSERT(resampledResponseLength == ResponseFrameSize);
+    } else {
+        resampled.SetLength(resampledResponseLength);
+        resampledResponse = resampled.Elements();
+        speex_resampler_skip_zeros(resampler);
+
+        // Feed the input buffer into the resampler.
+        spx_uint32_t in_len = ResponseFrameSize;
+        spx_uint32_t out_len = resampled.Length();
+        RESAMPLER_PROCESS(resampler, 0, response, &in_len,
+                          resampled.Elements(), &out_len);
+
+        if (out_len < resampled.Length()) {
+            // The input should have all been processed.
+            MOZ_ASSERT(in_len == ResponseFrameSize);
+            // Feed in zeros get the data remaining in the resampler.
+            spx_uint32_t out_index = out_len;
+            in_len = speex_resampler_get_input_latency(resampler);
+            out_len = resampled.Length() - out_index;
+            RESAMPLER_PROCESS(resampler, 0, nullptr, &in_len,
+                              resampled.Elements() + out_index, &out_len);
+            out_index += out_len;
+            // There may be some uninitialized samples remaining for very low
+            // sample rates.
+            PodZero(resampled.Elements() + out_index,
+                    resampled.Length() - out_index);
+        }
+
+        speex_resampler_reset_mem(resampler);
+    }
+
+#ifdef MOZ_SAMPLE_TYPE_S16
+    AutoTArray<float, 2 * ResponseFrameSize> floatArray;
+    floatArray.SetLength(resampledResponseLength);
+    float *floatResponse = floatArray.Elements();
+    ConvertAudioSamples(resampledResponse,
+                        floatResponse, resampledResponseLength);
+#else
+    float *floatResponse = resampledResponse;
+#endif
+#undef RESAMPLER_PROCESS
+
+    return HRTFKernel::create(floatResponse, resampledResponseLength, sampleRate);
+}
+
+// The range of elevations for the IRCAM impulse responses varies depending on azimuth, but the minimum elevation appears to always be -45.
+//
+// Here's how it goes:
+static int maxElevations[] = {
+        //  Azimuth
+        //
+    90, // 0  
+    45, // 15 
+    60, // 30 
+    45, // 45 
+    75, // 60 
+    45, // 75 
+    60, // 90 
+    45, // 105 
+    75, // 120 
+    45, // 135 
+    60, // 150 
+    45, // 165 
+    75, // 180 
+    45, // 195 
+    60, // 210 
+    45, // 225 
+    75, // 240 
+    45, // 255 
+    60, // 270 
+    45, // 285 
+    75, // 300 
+    45, // 315 
+    60, // 330 
+    45 //  345 
+};
+
+nsReturnRef<HRTFElevation> HRTFElevation::createBuiltin(int elevation, float sampleRate)
+{
+    if (elevation < firstElevation ||
+        elevation > firstElevation + numberOfElevations * elevationSpacing ||
+        (elevation / elevationSpacing) * elevationSpacing != elevation)
+        return nsReturnRef<HRTFElevation>();
+        
+    // Spacing, in degrees, between every azimuth loaded from resource.
+    // Some elevations do not have data for all these intervals.
+    // See maxElevations.
+    static const unsigned AzimuthSpacing = 15;
+    static const unsigned NumberOfRawAzimuths = 360 / AzimuthSpacing;
+    static_assert(AzimuthSpacing * NumberOfRawAzimuths == 360,
+                  "Not a multiple");
+    static const unsigned InterpolationFactor =
+        NumberOfTotalAzimuths / NumberOfRawAzimuths;
+    static_assert(NumberOfTotalAzimuths ==
+                  NumberOfRawAzimuths * InterpolationFactor, "Not a multiple");
+
+    HRTFKernelList kernelListL;
+    kernelListL.SetLength(NumberOfTotalAzimuths);
+
+    SpeexResamplerState* resampler = sampleRate == rawSampleRate ? nullptr :
+        speex_resampler_init(1, rawSampleRate, sampleRate,
+                             SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
+
+    // Load convolution kernels from HRTF files.
+    int interpolatedIndex = 0;
+    for (unsigned rawIndex = 0; rawIndex < NumberOfRawAzimuths; ++rawIndex) {
+        // Don't let elevation exceed maximum for this azimuth.
+        int maxElevation = maxElevations[rawIndex];
+        int actualElevation = min(elevation, maxElevation);
+
+        kernelListL[interpolatedIndex] = calculateKernelForAzimuthElevation(rawIndex * AzimuthSpacing, actualElevation, resampler, sampleRate);
+            
+        interpolatedIndex += InterpolationFactor;
+    }
+
+    if (resampler)
+        speex_resampler_destroy(resampler);
+
+    // Now go back and interpolate intermediate azimuth values.
+    for (unsigned i = 0; i < NumberOfTotalAzimuths; i += InterpolationFactor) {
+        int j = (i + InterpolationFactor) % NumberOfTotalAzimuths;
+
+        // Create the interpolated convolution kernels and delays.
+        for (unsigned jj = 1; jj < InterpolationFactor; ++jj) {
+            float x = float(jj) / float(InterpolationFactor); // interpolate from 0 -> 1
+
+            kernelListL[i + jj] = HRTFKernel::createInterpolatedKernel(kernelListL[i], kernelListL[j], x);
+        }
+    }
+    
+    return nsReturnRef<HRTFElevation>(new HRTFElevation(&kernelListL, elevation, sampleRate));
+}
+
+nsReturnRef<HRTFElevation> HRTFElevation::createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate)
+{
+    MOZ_ASSERT(hrtfElevation1 && hrtfElevation2);
+    if (!hrtfElevation1 || !hrtfElevation2)
+        return nsReturnRef<HRTFElevation>();
+        
+    MOZ_ASSERT(x >= 0.0 && x < 1.0);
+    
+    HRTFKernelList kernelListL;
+    kernelListL.SetLength(NumberOfTotalAzimuths);
+
+    const HRTFKernelList& kernelListL1 = hrtfElevation1->kernelListL();
+    const HRTFKernelList& kernelListL2 = hrtfElevation2->kernelListL();
+    
+    // Interpolate kernels of corresponding azimuths of the two elevations.
+    for (unsigned i = 0; i < NumberOfTotalAzimuths; ++i) {
+        kernelListL[i] = HRTFKernel::createInterpolatedKernel(kernelListL1[i], kernelListL2[i], x);
+    }
+
+    // Interpolate elevation angle.
+    double angle = (1.0 - x) * hrtfElevation1->elevationAngle() + x * hrtfElevation2->elevationAngle();
+    
+    return nsReturnRef<HRTFElevation>(new HRTFElevation(&kernelListL, static_cast<int>(angle), sampleRate));
+}
+
+void HRTFElevation::getKernelsFromAzimuth(double azimuthBlend, unsigned azimuthIndex, HRTFKernel* &kernelL, HRTFKernel* &kernelR, double& frameDelayL, double& frameDelayR)
+{
+    bool checkAzimuthBlend = azimuthBlend >= 0.0 && azimuthBlend < 1.0;
+    MOZ_ASSERT(checkAzimuthBlend);
+    if (!checkAzimuthBlend)
+        azimuthBlend = 0.0;
+    
+    unsigned numKernels = m_kernelListL.Length();
+
+    bool isIndexGood = azimuthIndex < numKernels;
+    MOZ_ASSERT(isIndexGood);
+    if (!isIndexGood) {
+        kernelL = 0;
+        kernelR = 0;
+        return;
+    }
+    
+    // Return the left and right kernels,
+    // using symmetry to produce the right kernel.
+    kernelL = m_kernelListL[azimuthIndex];
+    int azimuthIndexR = (numKernels - azimuthIndex) % numKernels;
+    kernelR = m_kernelListL[azimuthIndexR];
+
+    frameDelayL = kernelL->frameDelay();
+    frameDelayR = kernelR->frameDelay();
+
+    int azimuthIndex2L = (azimuthIndex + 1) % numKernels;
+    double frameDelay2L = m_kernelListL[azimuthIndex2L]->frameDelay();
+    int azimuthIndex2R = (numKernels - azimuthIndex2L) % numKernels;
+    double frameDelay2R = m_kernelListL[azimuthIndex2R]->frameDelay();
+
+    // Linearly interpolate delays.
+    frameDelayL = (1.0 - azimuthBlend) * frameDelayL + azimuthBlend * frameDelay2L;
+    frameDelayR = (1.0 - azimuthBlend) * frameDelayR + azimuthBlend * frameDelay2R;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFElevation.h b/dom/media/webaudio/blink/HRTFElevation.h
new file mode 100644
index 0000000000..e50947b12b
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFElevation.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFElevation_h
+#define HRTFElevation_h
+
+#include "HRTFKernel.h"
+#include "nsAutoRef.h"
+#include "mozilla/MemoryReporting.h"
+
+struct SpeexResamplerState_;
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+namespace WebCore {
+
+// HRTFElevation contains all of the HRTFKernels (one left ear and one right ear per azimuth angle) for a particular elevation.
+
+class HRTFElevation {
+public:
+    // Loads and returns an HRTFElevation with the given HRTF database subject name and elevation from browser (or WebKit.framework) resources.
+    // Normally, there will only be a single HRTF database set, but this API supports the possibility of multiple ones with different names.
+    // Interpolated azimuths will be generated based on InterpolationFactor.
+    // Valid values for elevation are -45 -> +90 in 15 degree increments.
+    static nsReturnRef<HRTFElevation> createBuiltin(int elevation, float sampleRate);
+
+    // Given two HRTFElevations, and an interpolation factor x: 0 -> 1, returns an interpolated HRTFElevation.
+    static nsReturnRef<HRTFElevation> createByInterpolatingSlices(HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x, float sampleRate);
+
+    double elevationAngle() const { return m_elevationAngle; }
+    unsigned numberOfAzimuths() const { return NumberOfTotalAzimuths; }
+    float sampleRate() const { return m_sampleRate; }
+    
+    // Returns the left and right kernels for the given azimuth index.
+    // The interpolated delays based on azimuthBlend: 0 -> 1 are returned in frameDelayL and frameDelayR.
+    void getKernelsFromAzimuth(double azimuthBlend, unsigned azimuthIndex, HRTFKernel* &kernelL, HRTFKernel* &kernelR, double& frameDelayL, double& frameDelayR);
+    
+    // Total number of azimuths after interpolation.
+    static const unsigned NumberOfTotalAzimuths;
+
+    static size_t fftSizeForSampleRate(float sampleRate);
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    HRTFElevation(const HRTFElevation& other) = delete;
+    void operator=(const HRTFElevation& other) = delete;
+
+    HRTFElevation(HRTFKernelList *kernelListL, int elevation, float sampleRate)
+        : m_elevationAngle(elevation)
+        , m_sampleRate(sampleRate)
+    {
+        m_kernelListL.SwapElements(*kernelListL);
+    }
+
+    // Returns the list of left ear HRTFKernels for all the azimuths going from 0 to 360 degrees.
+    const HRTFKernelList& kernelListL() { return m_kernelListL; }
+
+    // Given a specific azimuth and elevation angle, returns the left HRTFKernel.
+    // Values for azimuth must be multiples of 15 in 0 -> 345,
+    // but not all azimuths are available for elevations > +45.
+    // Valid values for elevation are -45 -> +90 in 15 degree increments.
+    static nsReturnRef<HRTFKernel> calculateKernelForAzimuthElevation(int azimuth, int elevation, SpeexResamplerState* resampler, float sampleRate);
+
+    HRTFKernelList m_kernelListL;
+    double m_elevationAngle;
+    float m_sampleRate;
+};
+
+} // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFElevation> :
+    public nsPointerRefTraits<WebCore::HRTFElevation> {
+public:
+    static void Release(WebCore::HRTFElevation* ptr) { delete(ptr); }
+};
+
+#endif // HRTFElevation_h
diff --git a/dom/media/webaudio/blink/HRTFKernel.cpp b/dom/media/webaudio/blink/HRTFKernel.cpp
new file mode 100644
index 0000000000..3ee5e63a36
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFKernel.cpp
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFKernel.h"
+namespace WebCore {
+
+// Takes the input audio channel |impulseP| as an input impulse response and calculates the average group delay.
+// This represents the initial delay before the most energetic part of the impulse response.
+// The sample-frame delay is removed from the |impulseP| impulse response, and this value  is returned.
+// The |length| of the passed in |impulseP| must be must be a power of 2.
+static float extractAverageGroupDelay(float* impulseP, size_t length)
+{
+    // Check for power-of-2.
+    MOZ_ASSERT(length && (length & (length - 1)) == 0);
+
+    FFTBlock estimationFrame(length);
+    estimationFrame.PerformFFT(impulseP);
+
+    float frameDelay = static_cast<float>(estimationFrame.ExtractAverageGroupDelay());
+    estimationFrame.GetInverse(impulseP);
+
+    return frameDelay;
+}
+
+HRTFKernel::HRTFKernel(float* impulseResponse, size_t length, float sampleRate)
+    : m_frameDelay(0)
+    , m_sampleRate(sampleRate)
+{
+    AlignedTArray<float> buffer;
+    // copy to a 32-byte aligned buffer
+    if (((uintptr_t)impulseResponse & 31) != 0) {
+      buffer.SetLength(length);
+      mozilla::PodCopy(buffer.Elements(), impulseResponse, length);
+      impulseResponse = buffer.Elements();
+    }
+
+    // Determine the leading delay (average group delay) for the response.
+    m_frameDelay = extractAverageGroupDelay(impulseResponse, length);
+
+    // The FFT size (with zero padding) needs to be twice the response length
+    // in order to do proper convolution.
+    unsigned fftSize = 2 * length;
+
+    // Quick fade-out (apply window) at truncation point
+    // because the impulse response has been truncated.
+    unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate
+    MOZ_ASSERT(numberOfFadeOutFrames < length);
+    if (numberOfFadeOutFrames < length) {
+        for (unsigned i = length - numberOfFadeOutFrames; i < length; ++i) {
+            float x = 1.0f - static_cast<float>(i - (length - numberOfFadeOutFrames)) / numberOfFadeOutFrames;
+            impulseResponse[i] *= x;
+        }
+    }
+
+    m_fftFrame = new FFTBlock(fftSize);
+    m_fftFrame->PadAndMakeScaledDFT(impulseResponse, length);
+}
+
+// Interpolates two kernels with x: 0 -> 1 and returns the result.
+nsReturnRef<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x)
+{
+    MOZ_ASSERT(kernel1 && kernel2);
+    if (!kernel1 || !kernel2)
+        return nsReturnRef<HRTFKernel>();
+
+    MOZ_ASSERT(x >= 0.0 && x < 1.0);
+    x = mozilla::clamped(x, 0.0f, 1.0f);
+
+    float sampleRate1 = kernel1->sampleRate();
+    float sampleRate2 = kernel2->sampleRate();
+    MOZ_ASSERT(sampleRate1 == sampleRate2);
+    if (sampleRate1 != sampleRate2)
+        return nsReturnRef<HRTFKernel>();
+
+    float frameDelay = (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();
+
+    nsAutoPtr<FFTBlock> interpolatedFrame(
+        FFTBlock::CreateInterpolatedBlock(*kernel1->fftFrame(), *kernel2->fftFrame(), x));
+    return HRTFKernel::create(interpolatedFrame, frameDelay, sampleRate1);
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFKernel.h b/dom/media/webaudio/blink/HRTFKernel.h
new file mode 100644
index 0000000000..940e69b139
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFKernel.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFKernel_h
+#define HRTFKernel_h
+
+#include "nsAutoPtr.h"
+#include "nsAutoRef.h"
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+using mozilla::FFTBlock;
+
+// HRTF stands for Head-Related Transfer Function.
+// HRTFKernel is a frequency-domain representation of an impulse-response used as part of the spatialized panning system.
+// For a given azimuth / elevation angle there will be one HRTFKernel for the left ear transfer function, and one for the right ear.
+// The leading delay (average group delay) for each impulse response is extracted:
+//      m_fftFrame is the frequency-domain representation of the impulse response with the delay removed
+//      m_frameDelay is the leading delay of the original impulse response.
+class HRTFKernel {
+public:
+    // Note: this is destructive on the passed in |impulseResponse|.
+    // The |length| of |impulseResponse| must be a power of two.
+    // The size of the DFT will be |2 * length|.
+    static nsReturnRef<HRTFKernel> create(float* impulseResponse, size_t length, float sampleRate);
+
+    static nsReturnRef<HRTFKernel> create(nsAutoPtr<FFTBlock> fftFrame, float frameDelay, float sampleRate);
+
+    // Given two HRTFKernels, and an interpolation factor x: 0 -> 1, returns an interpolated HRTFKernel.
+    static nsReturnRef<HRTFKernel> createInterpolatedKernel(HRTFKernel* kernel1, HRTFKernel* kernel2, float x);
+  
+    FFTBlock* fftFrame() { return m_fftFrame.get(); }
+    
+    size_t fftSize() const { return m_fftFrame->FFTSize(); }
+    float frameDelay() const { return m_frameDelay; }
+
+    float sampleRate() const { return m_sampleRate; }
+    double nyquist() const { return 0.5 * sampleRate(); }
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+    {
+        size_t amount = aMallocSizeOf(this);
+        amount += m_fftFrame->SizeOfIncludingThis(aMallocSizeOf);
+        return amount;
+    }
+
+private:
+    HRTFKernel(const HRTFKernel& other) = delete;
+    void operator=(const HRTFKernel& other) = delete;
+
+    // Note: this is destructive on the passed in |impulseResponse|.
+    HRTFKernel(float* impulseResponse, size_t fftSize, float sampleRate);
+    
+    HRTFKernel(nsAutoPtr<FFTBlock> fftFrame, float frameDelay, float sampleRate)
+        : m_fftFrame(fftFrame)
+        , m_frameDelay(frameDelay)
+        , m_sampleRate(sampleRate)
+    {
+    }
+    
+    nsAutoPtr<FFTBlock> m_fftFrame;
+    float m_frameDelay;
+    float m_sampleRate;
+};
+
+typedef nsTArray<nsAutoRef<HRTFKernel> > HRTFKernelList;
+
+} // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFKernel> :
+    public nsPointerRefTraits<WebCore::HRTFKernel> {
+public:
+    static void Release(WebCore::HRTFKernel* ptr) { delete(ptr); }
+};
+
+namespace WebCore {
+
+inline nsReturnRef<HRTFKernel> HRTFKernel::create(float* impulseResponse, size_t length, float sampleRate)
+{
+    return nsReturnRef<HRTFKernel>(new HRTFKernel(impulseResponse, length, sampleRate));
+}
+
+inline nsReturnRef<HRTFKernel> HRTFKernel::create(nsAutoPtr<FFTBlock> fftFrame, float frameDelay, float sampleRate)
+{
+    return nsReturnRef<HRTFKernel>(new HRTFKernel(fftFrame, frameDelay, sampleRate));
+}
+
+} // namespace WebCore
+
+#endif // HRTFKernel_h
diff --git a/dom/media/webaudio/blink/HRTFPanner.cpp b/dom/media/webaudio/blink/HRTFPanner.cpp
new file mode 100644
index 0000000000..c97ce47670
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFPanner.cpp
@@ -0,0 +1,324 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFPanner.h"
+#include "HRTFDatabaseLoader.h"
+
+#include "FFTConvolver.h"
+#include "HRTFDatabase.h"
+#include "AudioBlock.h"
+
+using namespace std;
+using namespace mozilla;
+using dom::ChannelInterpretation;
+
+namespace WebCore {
+
+// The value of 2 milliseconds is larger than the largest delay which exists in any HRTFKernel from the default HRTFDatabase (0.0136 seconds).
+// We ASSERT the delay values used in process() with this value.
+const double MaxDelayTimeSeconds = 0.002;
+
+const int UninitializedAzimuth = -1;
+const unsigned RenderingQuantum = WEBAUDIO_BLOCK_SIZE;
+
+HRTFPanner::HRTFPanner(float sampleRate, already_AddRefed<HRTFDatabaseLoader> databaseLoader)
+    : m_databaseLoader(databaseLoader)
+    , m_sampleRate(sampleRate)
+    , m_crossfadeSelection(CrossfadeSelection1)
+    , m_azimuthIndex1(UninitializedAzimuth)
+    , m_azimuthIndex2(UninitializedAzimuth)
+    // m_elevation1 and m_elevation2 are initialized in pan()
+    , m_crossfadeX(0)
+    , m_crossfadeIncr(0)
+    , m_convolverL1(HRTFElevation::fftSizeForSampleRate(sampleRate))
+    , m_convolverR1(m_convolverL1.fftSize())
+    , m_convolverL2(m_convolverL1.fftSize())
+    , m_convolverR2(m_convolverL1.fftSize())
+    , m_delayLine(MaxDelayTimeSeconds * sampleRate, 1.0)
+{
+    MOZ_ASSERT(m_databaseLoader);
+    MOZ_COUNT_CTOR(HRTFPanner);
+}
+
+HRTFPanner::~HRTFPanner()
+{
+    MOZ_COUNT_DTOR(HRTFPanner);
+}
+
+size_t HRTFPanner::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    // NB: m_databaseLoader can be shared, so it is not measured here
+    amount += m_convolverL1.sizeOfExcludingThis(aMallocSizeOf);
+    amount += m_convolverR1.sizeOfExcludingThis(aMallocSizeOf);
+    amount += m_convolverL2.sizeOfExcludingThis(aMallocSizeOf);
+    amount += m_convolverR2.sizeOfExcludingThis(aMallocSizeOf);
+    amount += m_delayLine.SizeOfExcludingThis(aMallocSizeOf);
+
+    return amount;
+}
+
+void HRTFPanner::reset()
+{
+    m_azimuthIndex1 = UninitializedAzimuth;
+    m_azimuthIndex2 = UninitializedAzimuth;
+    // m_elevation1 and m_elevation2 are initialized in pan()
+    m_crossfadeSelection = CrossfadeSelection1;
+    m_crossfadeX = 0.0f;
+    m_crossfadeIncr = 0.0f;
+    m_convolverL1.reset();
+    m_convolverR1.reset();
+    m_convolverL2.reset();
+    m_convolverR2.reset();
+    m_delayLine.Reset();
+}
+
+int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend)
+{
+    // Convert the azimuth angle from the range -180 -> +180 into the range 0 -> 360.
+    // The azimuth index may then be calculated from this positive value.
+    if (azimuth < 0)
+        azimuth += 360.0;
+
+    HRTFDatabase* database = m_databaseLoader->database();
+    MOZ_ASSERT(database);
+
+    int numberOfAzimuths = database->numberOfAzimuths();
+    const double angleBetweenAzimuths = 360.0 / numberOfAzimuths;
+
+    // Calculate the azimuth index and the blend (0 -> 1) for interpolation.
+    double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths;
+    int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat);
+    azimuthBlend = desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex);
+
+    // We don't immediately start using this azimuth index, but instead approach this index from the last index we rendered at.
+    // This minimizes the clicks and graininess for moving sources which occur otherwise.
+    desiredAzimuthIndex = max(0, desiredAzimuthIndex);
+    desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex);
+    return desiredAzimuthIndex;
+}
+
+void HRTFPanner::pan(double desiredAzimuth, double elevation, const AudioBlock* inputBus, AudioBlock* outputBus)
+{
+#ifdef DEBUG
+    unsigned numInputChannels =
+        inputBus->IsNull() ? 0 : inputBus->ChannelCount();
+
+    MOZ_ASSERT(numInputChannels <= 2);
+    MOZ_ASSERT(inputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE);
+#endif
+
+    bool isOutputGood = outputBus && outputBus->ChannelCount() == 2 && outputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE;
+    MOZ_ASSERT(isOutputGood);
+
+    if (!isOutputGood) {
+        if (outputBus)
+            outputBus->SetNull(outputBus->GetDuration());
+        return;
+    }
+
+    HRTFDatabase* database = m_databaseLoader->database();
+    if (!database) { // not yet loaded
+        outputBus->SetNull(outputBus->GetDuration());
+        return;
+    }
+
+    // IRCAM HRTF azimuths values from the loaded database is reversed from the panner's notion of azimuth.
+    double azimuth = -desiredAzimuth;
+
+    bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0;
+    MOZ_ASSERT(isAzimuthGood);
+    if (!isAzimuthGood) {
+        outputBus->SetNull(outputBus->GetDuration());
+        return;
+    }
+
+    // Normally, we'll just be dealing with mono sources.
+    // If we have a stereo input, implement stereo panning with left source processed by left HRTF, and right source by right HRTF.
+
+    // Get destination pointers.
+    float* destinationL =
+        static_cast<float*>(const_cast<void*>(outputBus->mChannelData[0]));
+    float* destinationR =
+        static_cast<float*>(const_cast<void*>(outputBus->mChannelData[1]));
+
+    double azimuthBlend;
+    int desiredAzimuthIndex = calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
+
+    // Initially snap azimuth and elevation values to first values encountered.
+    if (m_azimuthIndex1 == UninitializedAzimuth) {
+        m_azimuthIndex1 = desiredAzimuthIndex;
+        m_elevation1 = elevation;
+    }
+    if (m_azimuthIndex2 == UninitializedAzimuth) {
+        m_azimuthIndex2 = desiredAzimuthIndex;
+        m_elevation2 = elevation;
+    }
+
+    // Cross-fade / transition over a period of around 45 milliseconds.
+    // This is an empirical value tuned to be a reasonable trade-off between
+    // smoothness and speed.
+    const double fadeFrames = sampleRate() <= 48000 ? 2048 : 4096;
+
+    // Check for azimuth and elevation changes, initiating a cross-fade if needed.
+    if (!m_crossfadeX && m_crossfadeSelection == CrossfadeSelection1) {
+        if (desiredAzimuthIndex != m_azimuthIndex1 || elevation != m_elevation1) {
+            // Cross-fade from 1 -> 2
+            m_crossfadeIncr = 1 / fadeFrames;
+            m_azimuthIndex2 = desiredAzimuthIndex;
+            m_elevation2 = elevation;
+        }
+    }
+    if (m_crossfadeX == 1 && m_crossfadeSelection == CrossfadeSelection2) {
+        if (desiredAzimuthIndex != m_azimuthIndex2 || elevation != m_elevation2) {
+            // Cross-fade from 2 -> 1
+            m_crossfadeIncr = -1 / fadeFrames;
+            m_azimuthIndex1 = desiredAzimuthIndex;
+            m_elevation1 = elevation;
+        }
+    }
+
+    // Get the HRTFKernels and interpolated delays.
+    HRTFKernel* kernelL1;
+    HRTFKernel* kernelR1;
+    HRTFKernel* kernelL2;
+    HRTFKernel* kernelR2;
+    double frameDelayL1;
+    double frameDelayR1;
+    double frameDelayL2;
+    double frameDelayR2;
+    database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex1, m_elevation1, kernelL1, kernelR1, frameDelayL1, frameDelayR1);
+    database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex2, m_elevation2, kernelL2, kernelR2, frameDelayL2, frameDelayR2);
+
+    bool areKernelsGood = kernelL1 && kernelR1 && kernelL2 && kernelR2;
+    MOZ_ASSERT(areKernelsGood);
+    if (!areKernelsGood) {
+        outputBus->SetNull(outputBus->GetDuration());
+        return;
+    }
+
+    MOZ_ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds && frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
+    MOZ_ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds && frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
+
+    // Crossfade inter-aural delays based on transitions.
+    double frameDelaysL[WEBAUDIO_BLOCK_SIZE];
+    double frameDelaysR[WEBAUDIO_BLOCK_SIZE];
+    {
+      float x = m_crossfadeX;
+      float incr = m_crossfadeIncr;
+      for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+        frameDelaysL[i] = (1 - x) * frameDelayL1 + x * frameDelayL2;
+        frameDelaysR[i] = (1 - x) * frameDelayR1 + x * frameDelayR2;
+        x += incr;
+      }
+    }
+
+    // First run through delay lines for inter-aural time difference.
+    m_delayLine.Write(*inputBus);
+    // "Speakers" means a mono input is read into both outputs (with possibly
+    // different delays).
+    m_delayLine.ReadChannel(frameDelaysL, outputBus, 0,
+                            ChannelInterpretation::Speakers);
+    m_delayLine.ReadChannel(frameDelaysR, outputBus, 1,
+                            ChannelInterpretation::Speakers);
+    m_delayLine.NextBlock();
+
+    bool needsCrossfading = m_crossfadeIncr;
+
+    const float* convolutionDestinationL1;
+    const float* convolutionDestinationR1;
+    const float* convolutionDestinationL2;
+    const float* convolutionDestinationR2;
+
+    // Now do the convolutions.
+    // Note that we avoid doing convolutions on both sets of convolvers if we're not currently cross-fading.
+
+    if (m_crossfadeSelection == CrossfadeSelection1 || needsCrossfading) {
+        convolutionDestinationL1 =
+            m_convolverL1.process(kernelL1->fftFrame(), destinationL);
+        convolutionDestinationR1 =
+            m_convolverR1.process(kernelR1->fftFrame(), destinationR);
+    }
+
+    if (m_crossfadeSelection == CrossfadeSelection2 || needsCrossfading) {
+        convolutionDestinationL2 =
+            m_convolverL2.process(kernelL2->fftFrame(), destinationL);
+        convolutionDestinationR2 =
+            m_convolverR2.process(kernelR2->fftFrame(), destinationR);
+    }
+
+    if (needsCrossfading) {
+        // Apply linear cross-fade.
+        float x = m_crossfadeX;
+        float incr = m_crossfadeIncr;
+        for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+            destinationL[i] = (1 - x) * convolutionDestinationL1[i] + x * convolutionDestinationL2[i];
+            destinationR[i] = (1 - x) * convolutionDestinationR1[i] + x * convolutionDestinationR2[i];
+            x += incr;
+        }
+        // Update cross-fade value from local.
+        m_crossfadeX = x;
+
+        if (m_crossfadeIncr > 0 && fabs(m_crossfadeX - 1) < m_crossfadeIncr) {
+            // We've fully made the crossfade transition from 1 -> 2.
+            m_crossfadeSelection = CrossfadeSelection2;
+            m_crossfadeX = 1;
+            m_crossfadeIncr = 0;
+        } else if (m_crossfadeIncr < 0 && fabs(m_crossfadeX) < -m_crossfadeIncr) {
+            // We've fully made the crossfade transition from 2 -> 1.
+            m_crossfadeSelection = CrossfadeSelection1;
+            m_crossfadeX = 0;
+            m_crossfadeIncr = 0;
+        }
+    } else {
+        const float* sourceL;
+        const float* sourceR;
+        if (m_crossfadeSelection == CrossfadeSelection1) {
+            sourceL = convolutionDestinationL1;
+            sourceR = convolutionDestinationR1;
+        } else {
+            sourceL = convolutionDestinationL2;
+            sourceR = convolutionDestinationR2;
+        }
+        PodCopy(destinationL, sourceL, WEBAUDIO_BLOCK_SIZE);
+        PodCopy(destinationR, sourceR, WEBAUDIO_BLOCK_SIZE);
+    }
+}
+
+int HRTFPanner::maxTailFrames() const
+{
+    // Although the ideal tail time would be the length of the impulse
+    // response, there is additional tail time from the approximations in the
+    // implementation.  Because HRTFPanner is implemented with a DelayKernel
+    // and a FFTConvolver, the tailTime of the HRTFPanner is the sum of the
+    // tailTime of the DelayKernel and the tailTime of the FFTConvolver.  The
+    // FFTs of the convolver are fftSize(), half of which is latency, but this
+    // is aligned with blocks and so is reduced by the one block which is
+    // processed immediately.
+    return m_delayLine.MaxDelayTicks() +
+        m_convolverL1.fftSize()/2 + m_convolverL1.latencyFrames();
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFPanner.h b/dom/media/webaudio/blink/HRTFPanner.h
new file mode 100644
index 0000000000..f56d0d4232
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFPanner.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFPanner_h
+#define HRTFPanner_h
+
+#include "FFTConvolver.h"
+#include "DelayBuffer.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+class AudioBlock;
+} // namespace mozilla
+
+namespace WebCore {
+
+typedef nsTArray<float> AudioFloatArray;
+
+class HRTFDatabaseLoader;
+
+using mozilla::AudioBlock;
+
+class HRTFPanner {
+public:
+    HRTFPanner(float sampleRate, already_AddRefed<HRTFDatabaseLoader> databaseLoader);
+    ~HRTFPanner();
+
+    // chunk durations must be 128
+    void pan(double azimuth, double elevation, const AudioBlock* inputBus, AudioBlock* outputBus);
+    void reset();
+
+    size_t fftSize() const { return m_convolverL1.fftSize(); }
+
+    float sampleRate() const { return m_sampleRate; }
+
+    int maxTailFrames() const;
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    // Given an azimuth angle in the range -180 -> +180, returns the corresponding azimuth index for the database,
+    // and azimuthBlend which is an interpolation value from 0 -> 1.
+    int calculateDesiredAzimuthIndexAndBlend(double azimuth, double& azimuthBlend);
+
+    RefPtr<HRTFDatabaseLoader> m_databaseLoader;
+
+    float m_sampleRate;
+
+    // We maintain two sets of convolvers for smooth cross-faded interpolations when
+    // then azimuth and elevation are dynamically changing.
+    // When the azimuth and elevation are not changing, we simply process with one of the two sets.
+    // Initially we use CrossfadeSelection1 corresponding to m_convolverL1 and m_convolverR1.
+    // Whenever the azimuth or elevation changes, a crossfade is initiated to transition
+    // to the new position. So if we're currently processing with CrossfadeSelection1, then
+    // we transition to CrossfadeSelection2 (and vice versa).
+    // If we're in the middle of a transition, then we wait until it is complete before
+    // initiating a new transition.
+
+    // Selects either the convolver set (m_convolverL1, m_convolverR1) or (m_convolverL2, m_convolverR2).
+    enum CrossfadeSelection {
+        CrossfadeSelection1,
+        CrossfadeSelection2
+    };
+
+    CrossfadeSelection m_crossfadeSelection;
+
+    // azimuth/elevation for CrossfadeSelection1.
+    int m_azimuthIndex1;
+    double m_elevation1;
+
+    // azimuth/elevation for CrossfadeSelection2.
+    int m_azimuthIndex2;
+    double m_elevation2;
+
+    // A crossfade value 0 <= m_crossfadeX <= 1.
+    float m_crossfadeX;
+
+    // Per-sample-frame crossfade value increment.
+    float m_crossfadeIncr;
+
+    FFTConvolver m_convolverL1;
+    FFTConvolver m_convolverR1;
+    FFTConvolver m_convolverL2;
+    FFTConvolver m_convolverR2;
+
+    mozilla::DelayBuffer m_delayLine;
+
+    AudioFloatArray m_tempL1;
+    AudioFloatArray m_tempR1;
+    AudioFloatArray m_tempL2;
+    AudioFloatArray m_tempR2;
+};
+
+} // namespace WebCore
+
+#endif // HRTFPanner_h
diff --git a/dom/media/webaudio/blink/IIRFilter.cpp b/dom/media/webaudio/blink/IIRFilter.cpp
new file mode 100644
index 0000000000..94ec129c73
--- /dev/null
+++ b/dom/media/webaudio/blink/IIRFilter.cpp
@@ -0,0 +1,166 @@
+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "IIRFilter.h"
+
+#include <complex>
+
+namespace blink {
+
+// The length of the memory buffers for the IIR filter.  This MUST be a power of two and must be
+// greater than the possible length of the filter coefficients.
+const int kBufferLength = 32;
+static_assert(kBufferLength >= IIRFilter::kMaxOrder + 1,
+    "Internal IIR buffer length must be greater than maximum IIR Filter order.");
+
+IIRFilter::IIRFilter(const AudioDoubleArray* feedforwardCoef, const AudioDoubleArray* feedbackCoef)
+    : m_bufferIndex(0)
+    , m_feedback(feedbackCoef)
+    , m_feedforward(feedforwardCoef)
+{
+    m_xBuffer.SetLength(kBufferLength);
+    m_yBuffer.SetLength(kBufferLength);
+    reset();
+}
+
+IIRFilter::~IIRFilter()
+{
+}
+
+void IIRFilter::reset()
+{
+    memset(m_xBuffer.Elements(), 0, m_xBuffer.Length() * sizeof(double));
+    memset(m_yBuffer.Elements(), 0, m_yBuffer.Length() * sizeof(double));
+}
+
+static std::complex<double> evaluatePolynomial(const double* coef, std::complex<double> z, int order)
+{
+    // Use Horner's method to evaluate the polynomial P(z) = sum(coef[k]*z^k, k, 0, order);
+    std::complex<double> result = 0;
+
+    for (int k = order; k >= 0; --k)
+        result = result * z + std::complex<double>(coef[k]);
+
+    return result;
+}
+
+void IIRFilter::process(const float* sourceP, float* destP, size_t framesToProcess)
+{
+    // Compute
+    //
+    //   y[n] = sum(b[k] * x[n - k], k = 0, M) - sum(a[k] * y[n - k], k = 1, N)
+    //
+    // where b[k] are the feedforward coefficients and a[k] are the feedback coefficients of the
+    // filter.
+
+    // This is a Direct Form I implementation of an IIR Filter.  Should we consider doing a
+    // different implementation such as Transposed Direct Form II?
+    const double* feedback = m_feedback->Elements();
+    const double* feedforward = m_feedforward->Elements();
+
+    MOZ_ASSERT(feedback);
+    MOZ_ASSERT(feedforward);
+
+    // Sanity check to see if the feedback coefficients have been scaled appropriately. It must
+    // be EXACTLY 1!
+    MOZ_ASSERT(feedback[0] == 1);
+
+    int feedbackLength = m_feedback->Length();
+    int feedforwardLength = m_feedforward->Length();
+    int minLength = std::min(feedbackLength, feedforwardLength);
+
+    double* xBuffer = m_xBuffer.Elements();
+    double* yBuffer = m_yBuffer.Elements();
+
+    for (size_t n = 0; n < framesToProcess; ++n) {
+        // To help minimize roundoff, we compute using double's, even though the filter coefficients
+        // only have single precision values.
+        double yn = feedforward[0] * sourceP[n];
+
+        // Run both the feedforward and feedback terms together, when possible.
+        for (int k = 1; k < minLength; ++k) {
+            int n = (m_bufferIndex - k) & (kBufferLength - 1);
+            yn += feedforward[k] * xBuffer[n];
+            yn -= feedback[k] * yBuffer[n];
+        }
+
+        // Handle any remaining feedforward or feedback terms.
+        for (int k = minLength; k < feedforwardLength; ++k)
+            yn += feedforward[k] * xBuffer[(m_bufferIndex - k) & (kBufferLength - 1)];
+
+        for (int k = minLength; k < feedbackLength; ++k)
+            yn -= feedback[k] * yBuffer[(m_bufferIndex - k) & (kBufferLength - 1)];
+
+        // Save the current input and output values in the memory buffers for the next output.
+        m_xBuffer[m_bufferIndex] = sourceP[n];
+        m_yBuffer[m_bufferIndex] = yn;
+
+        m_bufferIndex = (m_bufferIndex + 1) & (kBufferLength - 1);
+
+        // Avoid introducing a stream of subnormals
+        // TODO: Remove this code when Bug 1157635 is fixed.
+        if (fabs(yn) >= FLT_MIN) {
+            destP[n] = yn;
+        } else {
+            destP[n] = 0.0;
+        }
+    }
+}
+
+void IIRFilter::getFrequencyResponse(int nFrequencies, const float* frequency, float* magResponse, float* phaseResponse)
+{
+    // Evaluate the z-transform of the filter at the given normalized frequencies from 0 to 1. (One
+    // corresponds to the Nyquist frequency.)
+    //
+    // The z-tranform of the filter is
+    //
+    // H(z) = sum(b[k]*z^(-k), k, 0, M) / sum(a[k]*z^(-k), k, 0, N);
+    //
+    // The desired frequency response is H(exp(j*omega)), where omega is in [0, 1).
+    //
+    // Let P(x) = sum(c[k]*x^k, k, 0, P) be a polynomial of order P.  Then each of the sums in H(z)
+    // is equivalent to evaluating a polynomial at the point 1/z.
+
+    for (int k = 0; k < nFrequencies; ++k) {
+        // zRecip = 1/z = exp(-j*frequency)
+        double omega = -M_PI * frequency[k];
+        std::complex<double> zRecip = std::complex<double>(cos(omega), sin(omega));
+
+        std::complex<double> numerator = evaluatePolynomial(m_feedforward->Elements(), zRecip, m_feedforward->Length() - 1);
+        std::complex<double> denominator = evaluatePolynomial(m_feedback->Elements(), zRecip, m_feedback->Length() - 1);
+        // Strangely enough, using complex division:
+        // e.g. Complex response = numerator / denominator;
+        // fails on our test machines, yielding infinities and NaNs, so we do
+        // things the long way here.
+        double n = norm(denominator);
+        double r = (real(numerator)*real(denominator) + imag(numerator)*imag(denominator)) / n;
+        double i = (imag(numerator)*real(denominator) - real(numerator)*imag(denominator)) / n;
+        std::complex<double> response = std::complex<double>(r, i);
+
+        magResponse[k] = static_cast<float>(abs(response));
+        phaseResponse[k] = static_cast<float>(atan2(imag(response), real(response)));
+    }
+}
+
+bool IIRFilter::buffersAreZero()
+{
+    double* xBuffer = m_xBuffer.Elements();
+    double* yBuffer = m_yBuffer.Elements();
+
+    for (size_t k = 0; k < m_feedforward->Length(); ++k) {
+        if (xBuffer[(m_bufferIndex - k) & (kBufferLength - 1)] != 0.0) {
+            return false;
+        }
+    }
+
+    for (size_t k = 0; k < m_feedback->Length(); ++k) {
+        if (fabs(yBuffer[(m_bufferIndex - k) & (kBufferLength - 1)]) >= FLT_MIN) {
+            return false;
+        }
+    }
+
+    return true;
+}
+
+} // namespace blink
diff --git a/dom/media/webaudio/blink/IIRFilter.h b/dom/media/webaudio/blink/IIRFilter.h
new file mode 100644
index 0000000000..5656d959a4
--- /dev/null
+++ b/dom/media/webaudio/blink/IIRFilter.h
@@ -0,0 +1,58 @@
+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef IIRFilter_h
+#define IIRFilter_h
+
+typedef nsTArray<double> AudioDoubleArray;
+
+namespace blink {
+
+class IIRFilter final {
+public:
+    // The maximum IIR filter order.  This also limits the number of feedforward coefficients.  The
+    // maximum number of coefficients is 20 according to the spec.
+    const static size_t kMaxOrder = 19;
+    IIRFilter(const AudioDoubleArray* feedforwardCoef, const AudioDoubleArray* feedbackCoef);
+    ~IIRFilter();
+
+    void process(const float* sourceP, float* destP, size_t framesToProcess);
+
+    void reset();
+
+    void getFrequencyResponse(int nFrequencies,
+        const float* frequency,
+        float* magResponse,
+        float* phaseResponse);
+
+    bool buffersAreZero();
+
+private:
+    // Filter memory
+    //
+    // For simplicity, we assume |m_xBuffer| and |m_yBuffer| have the same length, and the length is
+    // a power of two.  Since the number of coefficients has a fixed upper length, the size of
+    // xBuffer and yBuffer is fixed. |m_xBuffer| holds the old input values and |m_yBuffer| holds
+    // the old output values needed to compute the new output value.
+    //
+    // m_yBuffer[m_bufferIndex] holds the most recent output value, say, y[n].  Then
+    // m_yBuffer[m_bufferIndex - k] is y[n - k].  Similarly for m_xBuffer.
+    //
+    // To minimize roundoff, these arrays are double's instead of floats.
+    AudioDoubleArray m_xBuffer;
+    AudioDoubleArray m_yBuffer;
+
+    // Index into the xBuffer and yBuffer arrays where the most current x and y values should be
+    // stored.  xBuffer[bufferIndex] corresponds to x[n], the current x input value and
+    // yBuffer[bufferIndex] is where y[n], the current output value.
+    int m_bufferIndex;
+
+    // Coefficients of the IIR filter.
+    const AudioDoubleArray* m_feedback;
+    const AudioDoubleArray* m_feedforward;
+};
+
+} // namespace blink
+
+#endif // IIRFilter_h
diff --git a/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp b/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp
new file mode 100644
index 0000000000..daffb114fd
--- /dev/null
+++ b/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp
@@ -0,0 +1,449 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/**
+ * The sample data in the arrays here was derived for Webkit by Chris Rogers
+ * through averaging of impulse responses from the IRCAM Listen HRTF Database.
+ * The responses here are half the length of the Listen responses.  This
+ * sample data has been granted to the Public Domain.
+ *
+ * This file is intended to be included in compilation of a single
+ * implementation file.
+ *
+ * Each elevation (p) contains impulse responses at a varying number of
+ * equally spaced azimuths for the left ear, ordered clockwise from in front
+ * the listener.
+ */
+
+#include "mozilla/ArrayUtils.h"
+
+using mozilla::ArrayLength;
+
+const int16_t irc_composite_c_r0195_p315[][256] =
+  {/* IRC_Composite_C_R0195_T000_P315.wav */
+   {-37,37,-38,39,-39,40,-41,42,-42,43,-43,44,-44,44,-44,44,-43,42,-39,36,-31,23,-10,-10,5,-655,-410,-552,-353,-474,-1525,-758,656,-263,70,1414,-1528,-731,-1811,1646,1312,-1501,-407,8893,-1543,7696,8084,2629,-2452,-234,3799,1676,177,-1077,-474,-1325,3527,985,265,-884,373,971,1024,-412,507,-173,259,-799,635,-628,507,-344,394,-359,178,-276,349,-201,137,-249,377,-311,263,-404,284,-244,173,-243,330,-320,112,-150,164,-142,174,-300,158,-197,13,-141,85,-190,64,-122,41,-122,60,-195,125,-163,10,-67,-6,-122,77,-133,26,-71,-42,-156,48,-152,-12,-89,-120,-104,-37,-154,-57,-139,-80,-165,-95,-242,-81,-146,-111,-178,-109,-208,-48,-178,-131,-163,-68,-169,-94,-190,-139,-190,-118,-204,-160,-220,-140,-204,-171,-238,-126,-203,-114,-209,-138,-177,-124,-184,-130,-175,-170,-185,-180,-231,-189,-233,-210,-236,-245,-288,-208,-329,-246,-274,-199,-273,-189,-267,-208,-215,-199,-187,-209,-206,-210,-123,-197,-156,-173,-142,-97,-123,-97,-107,-73,-84,-39,-50,-66,-11,-50,-12,-51,8,-27,19,-48,-9,-18,5,-42,-15,-35,-31,-27,-27,-64,-33,-54,-1,-98,-47,-56,-7,-76,-47,-70,-42,-54,-65,-76,-43,-57,-9,-61,-39,-58,33,-39,3,-34,20,-19,4,-71,61,-22,10},
+   /* IRC_Composite_C_R0195_T015_P315.wav */
+   {81,-82,83,-84,84,-85,86,-87,87,-87,87,-87,87,-86,84,-82,78,-72,63,-49,23,33,-344,-481,-563,-443,-265,-1527,-713,251,-1453,939,2510,-1221,-1282,-1307,806,585,-990,-82,9029,-4621,9096,11230,4611,-3051,400,2105,749,1644,165,-1556,-1521,3009,1430,723,-902,933,187,28,480,951,-214,122,-730,-95,-137,573,-593,558,-692,-62,28,16,-505,426,-283,227,-320,210,-374,303,-435,127,-128,76,-349,106,-364,139,-348,184,-425,-36,-441,91,-413,93,-444,33,-257,-74,-414,93,-379,19,-327,-43,-366,79,-293,20,-199,-76,-207,149,-239,68,-247,66,-219,61,-232,38,-212,36,-209,-27,-209,-58,-227,-6,-309,-12,-225,-60,-199,-10,-277,19,-207,-69,-211,-30,-261,-24,-233,-102,-217,-131,-247,-144,-229,-111,-256,-104,-254,-130,-241,-66,-249,-88,-223,-144,-199,-148,-229,-101,-242,-189,-240,-163,-308,-147,-285,-217,-239,-224,-291,-173,-269,-220,-209,-208,-240,-180,-212,-201,-217,-169,-184,-174,-178,-146,-209,-108,-186,-96,-108,-90,-120,-44,-101,-49,-39,-22,-44,1,-58,3,-19,-21,-8,-6,-22,-9,18,-15,-7,-23,-12,-24,-11,-43,-33,-33,-31,-57,-5,-64,-22,-45,-16,-87,-8,-40,-45,-57,-14,-53,-5,-47,-21,-45,-8,-33,13,-16,3,-29,6,-18,23,-31,24,-41},
+   /* IRC_Composite_C_R0195_T030_P315.wav */
+   {9,-9,8,-8,8,-8,8,-7,7,-6,5,-4,3,-2,0,2,-5,9,-14,23,-127,-380,-922,-742,-548,257,-2362,972,-1653,1079,1193,1855,-423,-3573,-1945,3974,274,-2796,11656,-988,5085,9362,10293,-750,-3105,-1864,2111,2283,-318,-221,-1158,-110,2923,410,-981,900,122,-303,833,532,-152,20,-462,17,-48,336,235,63,-647,577,-362,-19,-259,102,168,-103,-338,-333,24,-9,-458,287,-470,-273,-216,46,-484,13,-220,-143,-392,-290,-340,-198,-301,-237,-241,-176,-489,-201,-271,-252,-335,-60,-395,-225,-412,-191,-198,-164,-259,-64,-380,-75,-287,-9,-159,-126,-100,-116,-155,-96,-113,-23,-100,-117,-115,-164,-180,8,-173,-57,-208,-89,-129,-73,-188,-102,-184,-93,-204,-81,-217,-128,-180,-66,-279,-127,-244,-206,-146,-178,-190,-135,-228,-154,-205,-100,-196,-143,-185,-140,-235,-135,-209,-144,-158,-250,-202,-234,-227,-157,-214,-201,-260,-180,-244,-184,-234,-151,-192,-198,-223,-199,-199,-202,-189,-164,-233,-143,-220,-175,-186,-107,-156,-84,-181,-106,-116,-72,-63,-29,-76,-11,-83,15,-65,30,-2,-4,-41,4,-49,38,-44,57,-2,9,-61,11,-29,35,-27,2,-50,-23,-17,1,-57,2,-26,-14,-67,-39,-9,-30,-30,-21,-44,-39,-75,-12,-27,-13,-32,13,-22,-16,5,-2,23,-25,48,-24,-32,-30},
+   /* IRC_Composite_C_R0195_T045_P315.wav */
+   {52,-54,57,-59,62,-64,67,-71,74,-78,82,-86,91,-95,100,-104,107,-105,80,-502,-332,-993,-162,-1907,1505,-2021,-314,756,-842,5057,-1148,-2263,-3462,3639,-1435,4503,-272,9367,494,8006,14353,933,-5033,-2453,2315,1073,2918,-1236,-3010,2416,371,1956,-531,-1294,-177,691,1104,-312,402,-1300,693,-586,315,-26,538,-594,294,-267,-119,-580,378,28,261,-1010,385,-342,8,-224,217,-399,43,-603,-1,-295,-252,167,-340,-419,-328,-348,-86,-497,-57,-294,-381,-499,-201,-454,-232,-518,-17,-717,-310,-542,-84,-458,-96,-313,-154,-496,-112,-226,-165,-242,-163,-234,-306,-270,-70,-238,-108,-139,-101,-251,-218,-175,-17,-215,-23,-205,-109,-221,-12,-131,-27,-205,-45,-264,-158,-151,-4,-169,-113,-251,-102,-247,-100,-151,-158,-212,-170,-190,-217,-220,-105,-182,-191,-216,-210,-208,-188,-144,-93,-288,-132,-248,-124,-281,-112,-217,-92,-321,-80,-343,-94,-222,-100,-231,-229,-216,-139,-209,-164,-138,-158,-204,-180,-171,-145,-125,-68,-150,-139,-133,-97,-43,-133,-83,-49,-109,-53,-101,-20,-84,23,-64,-10,-116,38,-50,15,-23,-2,-3,-27,-28,3,14,36,-13,-21,-18,-38,28,-26,9,-12,-28,-46,-29,-8,8,18,-21,-12,-47,16,-23,-21,-22,-28,-21,-46,-32,37,-14,-1,-31,11,-4,-17,51,-42,23,-30,55},
+   /* IRC_Composite_C_R0195_T060_P315.wav */
+   {-9,10,-10,10,-11,11,-11,12,-12,13,-13,14,-14,14,-15,15,-14,14,-237,-853,-211,-839,-537,-1171,1035,-1099,1039,294,-1596,6549,-2739,-2660,-4050,4749,2134,7195,4024,6882,-1377,13010,8996,-6905,-3319,-3088,4606,2892,2461,-4423,-1310,1787,1273,1672,-1868,-79,1190,70,-141,-131,222,-677,570,-820,675,-811,128,-382,165,-353,-183,-560,68,-440,-382,-163,-67,-467,-152,-570,177,-472,-46,-374,-58,-324,-179,-380,-114,-308,-223,-231,-266,-228,-188,-382,-93,-468,-172,-439,-277,-467,-233,-484,-158,-431,-177,-375,-153,-360,-208,-377,-90,-359,-252,-321,-261,-288,-236,-352,-117,-397,-126,-318,-103,-229,-203,-302,-132,-152,-113,-159,-188,-103,-160,-141,-121,-237,-158,-186,-215,-126,-180,-203,-129,-258,-131,-232,-90,-192,-139,-222,-58,-238,-81,-214,-111,-210,-119,-232,-97,-257,-79,-254,-124,-244,-169,-224,-170,-218,-187,-194,-208,-142,-212,-136,-216,-224,-142,-210,-181,-238,-144,-243,-81,-233,-107,-193,-82,-124,-113,-114,-86,-61,-72,-88,-74,-42,-62,-68,-57,-28,-20,-32,-42,-4,-55,-25,11,-34,-17,-21,12,-38,-17,-4,-21,-19,-51,-19,-40,-1,-58,-23,-65,-46,-5,-23,-26,-24,-22,36,-42,4,-20,73,-50,19,-61,15,-46,1,-36,-37,-6,-9,13,-34,18,-10,58,-13,38,32,29,4,16,-6},
+   /* IRC_Composite_C_R0195_T075_P315.wav */
+   {-13,13,-14,14,-14,14,-14,14,-14,13,-13,11,-9,5,2,-16,50,-741,214,-932,242,-1432,-236,-662,1347,-571,-1196,4105,-1805,3633,-3512,-1059,-3340,6144,8904,2949,-3056,13761,4639,6581,1016,-7994,-537,2069,9498,-3772,-2314,-3272,3945,434,437,-1200,-83,923,138,258,258,-7,455,-141,284,-478,398,-1090,528,-789,-29,-665,-287,-554,-73,-808,-317,-229,-409,-754,-201,-562,103,-767,-233,-393,90,-725,-54,-341,-112,-375,-320,-304,-39,-501,-232,-150,-220,-432,-164,-401,-81,-462,-293,-278,-139,-468,-172,-335,-180,-318,-233,-226,-103,-361,-214,-194,-168,-399,-129,-254,-92,-285,-156,-134,-28,-353,-136,-218,-68,-245,-164,-257,-128,-294,-188,-259,-264,-283,-216,-266,-192,-249,-243,-221,-193,-297,-159,-234,-247,-176,-224,-169,-180,-199,-220,-117,-192,-112,-197,-146,-176,-92,-190,-108,-218,-108,-210,-116,-237,-105,-215,-104,-222,-112,-224,-93,-273,-105,-266,-128,-293,-88,-283,-37,-279,-49,-211,-28,-227,20,-197,2,-189,32,-205,58,-176,59,-144,57,-167,132,-119,93,-104,104,-78,87,-109,101,-75,67,-98,89,-87,61,-85,56,-78,56,-103,66,-107,43,-101,53,-122,26,-89,31,-77,5,-71,25,-79,9,-93,21,-74,23,-83,51,-92,20,-72,56,-59,56,-36,38,19,39,4,62,-17,56,-23,68},
+   /* IRC_Composite_C_R0195_T090_P315.wav */
+   {87,-92,97,-103,109,-116,123,-131,140,-150,162,-174,189,-205,224,-244,266,-286,51,-859,535,-1364,139,-1195,1589,-390,911,-1048,4305,473,-5108,558,766,2725,10227,-95,2100,7357,10939,4034,-9033,1273,-876,7923,448,-413,-5710,1509,967,2067,-1395,-1318,853,624,-242,51,284,401,299,341,-114,602,-538,101,-460,-168,-515,-276,-591,-395,-537,-566,-206,-633,-470,-595,-421,-387,-497,-528,-554,-277,-415,-451,-399,-376,-198,-441,-287,-406,-190,-420,-196,-316,-218,-371,-230,-316,-251,-337,-304,-180,-217,-314,-203,-175,-197,-285,-220,-238,-109,-305,-167,-234,-146,-316,-92,-192,-70,-236,-136,-80,-80,-227,-175,-115,-77,-157,-162,-216,-61,-249,-168,-273,-182,-316,-207,-288,-178,-282,-287,-286,-206,-333,-257,-356,-226,-357,-253,-372,-144,-341,-201,-306,-120,-225,-145,-256,-69,-151,-103,-191,-78,-159,-72,-231,-86,-209,-102,-254,-75,-225,-131,-246,-122,-189,-129,-238,-123,-178,-85,-185,-80,-172,-78,-165,-42,-131,-62,-158,-35,-137,-15,-123,-23,-84,14,-68,23,-29,8,12,-7,21,17,-7,6,-12,18,-60,33,-67,51,-69,41,-34,41,-45,33,-37,31,-56,11,-29,-2,-76,4,-43,17,-73,-1,-63,19,-62,-23,-71,-23,-50,-30,-62,24,-59,23,-75,63,-53,35,-32,38,-9,16,4,53,-2,34,-15},
+   /* IRC_Composite_C_R0195_T105_P315.wav */
+   {-4,4,-4,4,-4,4,-4,4,-5,5,-5,6,-7,9,-11,17,-28,67,21,-444,-141,-224,-503,-492,56,581,659,-254,970,3243,-1085,-4435,2584,328,6386,5564,-809,5990,6762,10583,-5626,-2592,332,5216,4379,-2326,-2369,-2841,2364,794,414,-2336,841,434,185,-245,467,54,226,66,291,109,-14,-250,-37,-434,-55,-468,-301,-501,-528,-390,-260,-361,-561,-460,-370,-387,-354,-670,-414,-508,-325,-671,-240,-470,-287,-539,-267,-403,-309,-523,-407,-423,-444,-437,-347,-265,-394,-345,-296,-231,-221,-203,-113,-208,-173,-209,-71,-127,-120,-168,-119,-150,-164,-160,-171,-155,-129,-132,-133,-114,-84,-152,-145,-150,-50,-140,-100,-199,-121,-188,-145,-209,-221,-240,-266,-216,-273,-254,-311,-241,-309,-303,-306,-288,-304,-324,-325,-280,-286,-254,-285,-213,-282,-199,-235,-168,-198,-151,-152,-157,-138,-157,-155,-189,-176,-212,-179,-192,-145,-184,-171,-174,-110,-150,-113,-170,-84,-115,-76,-135,-71,-161,-62,-153,-49,-144,-73,-126,-80,-100,-82,-63,-82,-57,-74,14,-40,-10,-30,25,-21,31,-22,45,-26,43,-15,48,-18,20,-19,34,-32,26,-30,33,-39,33,-38,25,-62,21,-45,-12,-65,-6,-49,-13,-61,8,-71,9,-80,15,-87,37,-62,49,-58,33,-30,20,-31,-30,-19,-18,-22,12,-19,18,-18,62,-6,19},
+   /* IRC_Composite_C_R0195_T120_P315.wav */
+   {-8,8,-9,10,-10,11,-12,13,-14,15,-17,18,-20,22,-24,27,-30,33,-36,-6,-353,-241,59,-307,-656,81,435,931,-462,1098,2228,1185,-5297,2389,1039,4366,5054,563,5052,5065,10308,-3912,-1912,-658,6384,3010,-2420,-2503,-2288,1665,1300,-465,-1526,865,689,-244,192,664,-256,-73,265,-57,-312,-7,-322,-127,-354,-230,-494,-275,-378,-331,-253,-267,-433,-173,-259,-320,-294,-344,-567,-293,-505,-543,-515,-293,-398,-306,-402,-421,-360,-360,-466,-547,-514,-519,-327,-326,-417,-471,-219,-314,-177,-232,-161,-243,-135,-249,-154,-136,-89,-210,-64,-203,-123,-169,-90,-247,-100,-141,-68,-108,-41,-101,-97,-95,-145,-67,-56,-122,-193,-178,-180,-188,-181,-281,-271,-252,-234,-263,-302,-242,-343,-214,-391,-204,-407,-189,-457,-208,-383,-206,-352,-186,-336,-183,-268,-158,-243,-145,-229,-99,-205,-110,-223,-83,-256,-120,-241,-124,-258,-103,-238,-73,-205,-48,-194,-12,-168,-14,-161,-15,-200,-2,-184,-29,-168,-39,-203,-35,-143,-59,-138,-57,-120,-39,-78,-38,-71,4,-85,25,-41,42,-65,60,-35,71,-28,56,-41,66,-43,40,-26,-6,-23,-11,-1,-28,3,-16,-8,-25,-4,-23,-19,-21,-43,6,-38,-12,-49,4,-21,-1,-32,9,-8,22,-23,-1,-11,-14,-32,-12,-40,-9,-31,23,-39,25,-13,20,-8,9},
+   /* IRC_Composite_C_R0195_T135_P315.wav */
+   {-23,24,-24,24,-24,24,-24,24,-24,24,-24,24,-24,23,-23,22,-21,19,-17,12,2,-321,142,-352,-116,-71,-511,212,1241,-690,1132,1447,1818,-2325,-1459,2842,2322,3950,-401,7294,1799,9821,384,-3014,-458,4980,4067,-3265,-1817,-2557,1536,1050,141,-1408,467,945,477,-318,841,-225,-162,-191,333,-541,-289,-847,48,-224,-228,-453,-42,-332,-101,-393,-171,-405,-103,-444,-70,-591,-149,-459,-69,-570,-117,-584,-95,-396,-42,-496,-210,-651,-204,-618,-333,-581,-255,-540,-313,-471,-202,-369,-230,-385,-202,-278,-178,-259,-149,-205,-123,-195,-100,-150,-73,-188,-126,-225,-124,-193,-92,-182,-65,-172,-74,-168,-103,-186,-57,-178,-89,-280,-140,-220,-97,-251,-226,-268,-204,-268,-271,-286,-293,-290,-360,-269,-322,-274,-307,-261,-331,-275,-290,-250,-261,-307,-242,-239,-199,-225,-211,-141,-180,-148,-189,-147,-163,-178,-155,-195,-114,-193,-96,-201,-69,-171,-54,-158,-66,-130,-90,-112,-83,-108,-98,-108,-90,-89,-94,-103,-90,-117,-87,-105,-51,-97,-64,-70,-56,-23,-79,7,-88,34,-80,43,-77,63,-50,59,-68,43,-51,41,-54,60,-92,29,-102,48,-85,28,-79,9,-67,13,-66,37,-57,38,-64,47,-47,53,-38,54,-42,32,-33,27,-15,13,-24,-26,-34,-6,-41,23,-23,1,-31,23,-1,22,-48,-14},
+   /* IRC_Composite_C_R0195_T150_P315.wav */
+   {-14,14,-14,14,-14,14,-14,14,-14,14,-15,15,-15,15,-15,15,-15,15,-15,15,-14,13,-9,-299,105,-109,-149,-174,-101,123,928,331,409,1091,1957,-1219,-1800,2020,2641,4225,-365,6765,1276,7468,2095,-1421,-1941,2847,4183,-2838,-2013,-2585,1301,1369,1048,-1018,299,712,620,-358,419,-160,-391,-365,85,-244,-278,-735,101,-159,-46,-323,-112,-300,-73,-375,-177,-408,-241,-349,-232,-359,-171,-320,-131,-293,-2,-318,-62,-331,-78,-385,-166,-449,-212,-441,-266,-408,-251,-382,-246,-414,-254,-434,-274,-407,-251,-444,-252,-399,-189,-349,-176,-289,-151,-238,-106,-181,-45,-201,-17,-170,-8,-151,12,-150,-68,-196,-148,-187,-145,-232,-247,-286,-284,-266,-250,-295,-275,-287,-281,-302,-308,-288,-299,-267,-330,-280,-301,-267,-269,-255,-300,-276,-267,-240,-267,-229,-279,-186,-258,-159,-229,-176,-170,-148,-137,-189,-140,-182,-128,-193,-123,-144,-95,-99,-87,-75,-90,-47,-89,-71,-115,-103,-102,-112,-110,-131,-87,-120,-99,-104,-130,-68,-115,-50,-116,-33,-115,-23,-91,-20,-77,-32,-46,-36,-20,-19,-1,-1,-16,2,-30,-4,-10,2,-12,-25,-22,-46,-39,-40,-29,-51,-40,-76,-30,-47,-17,-48,-11,-52,8,-41,31,-26,41,1,54,-33,53,-2,35,-28,34,-23,25,-31,33,-25,-2,-7,10,-26,-60,-40},
+   /* IRC_Composite_C_R0195_T165_P315.wav */
+   {-17,17,-18,19,-19,20,-21,21,-22,23,-24,25,-26,27,-29,30,-32,33,-35,37,-39,42,-45,48,-52,67,-40,152,75,-119,-31,837,302,348,1063,1535,471,-104,-785,1551,1039,5803,545,3679,3043,5986,50,-1878,1830,-1451,2642,-2294,-1114,-2432,2146,1209,-135,-693,285,141,308,-50,-77,-237,196,-409,80,-447,-119,-467,-15,-415,-52,-429,-63,-216,-215,-423,-211,-250,-99,-343,-232,-231,-78,-380,-66,-225,10,-172,-61,-149,-128,-331,-281,-369,-348,-420,-336,-400,-303,-402,-320,-360,-377,-402,-338,-363,-356,-382,-392,-353,-371,-352,-344,-280,-280,-250,-233,-166,-154,-160,-136,-102,-71,-93,-69,-118,-111,-164,-179,-223,-230,-272,-255,-285,-312,-330,-270,-347,-306,-400,-338,-356,-324,-353,-315,-327,-313,-301,-321,-289,-309,-258,-317,-227,-285,-196,-252,-189,-237,-155,-182,-95,-165,-128,-123,-87,-127,-113,-150,-118,-122,-92,-114,-79,-90,-44,-70,-19,-82,-37,-76,-88,-84,-86,-94,-117,-109,-92,-108,-75,-136,-74,-119,-62,-120,-51,-126,-38,-97,-20,-110,-3,-105,29,-72,41,-66,43,-69,58,-54,55,-37,38,-45,18,-47,10,-73,-1,-104,15,-120,0,-103,12,-106,19,-73,20,-41,43,-9,42,-5,48,-4,46,3,29,-8,24,11,13,-8,11,55,13,32,16,31,9,-51,-42},
+   /* IRC_Composite_C_R0195_T180_P315.wav */
+   {24,-25,26,-27,28,-30,31,-33,34,-36,38,-40,42,-45,47,-50,53,-57,61,-65,70,-76,82,-90,100,-116,158,-120,110,-47,143,-200,288,-83,509,566,678,584,1717,-752,-218,-645,3620,1547,2553,2618,2850,3932,1154,2478,-3553,2655,-1012,521,-1943,-220,418,465,358,-445,32,572,410,189,111,242,-109,240,99,120,-178,-44,-146,-12,-251,-169,-110,-64,-56,-37,-65,38,137,-75,-42,-159,-127,-127,-11,-39,-7,-121,-97,-173,-207,-322,-357,-395,-311,-334,-329,-369,-303,-285,-259,-281,-323,-274,-364,-264,-415,-314,-459,-317,-463,-346,-399,-298,-323,-239,-217,-185,-167,-162,-146,-115,-142,-125,-194,-139,-196,-150,-224,-208,-224,-201,-206,-219,-254,-241,-239,-289,-306,-358,-320,-362,-312,-398,-353,-429,-362,-421,-352,-421,-332,-364,-272,-351,-188,-256,-150,-222,-109,-165,-57,-142,-72,-111,-54,-116,-66,-134,-87,-132,-39,-149,-55,-144,-3,-111,-37,-108,-57,-98,-89,-106,-109,-135,-117,-147,-114,-137,-99,-140,-85,-140,-56,-119,-68,-119,-57,-94,-53,-95,-61,-85,-42,-76,-22,-61,-39,-37,-44,-27,-31,-14,-37,-23,-16,-39,0,-62,-11,-92,2,-64,-27,-61,-33,-39,-34,-35,-26,-27,5,-20,14,-33,30,-28,12,-27,25,-7,7,29,-15,24,-9,67,32,52,-7,19,-10,-34},
+   /* IRC_Composite_C_R0195_T195_P315.wav */
+   {-3,3,-4,4,-4,4,-4,4,-4,4,-4,4,-5,5,-5,5,-5,5,-6,6,-6,7,-7,8,-10,16,-53,-21,71,-52,123,66,-67,138,16,149,471,552,491,879,834,185,-864,886,1624,2747,1377,2335,2675,2476,2941,-204,-424,-902,1080,-594,-652,-972,-15,737,158,15,-103,673,669,117,134,496,369,6,174,211,204,39,24,7,-16,8,-20,185,63,80,52,87,-18,-50,-107,-83,-58,-90,-149,-191,-215,-179,-280,-228,-426,-325,-403,-260,-377,-309,-454,-294,-358,-263,-350,-288,-312,-298,-376,-336,-392,-351,-440,-368,-423,-300,-356,-221,-309,-156,-237,-133,-233,-159,-209,-147,-233,-211,-247,-210,-236,-217,-261,-238,-270,-208,-309,-227,-327,-269,-364,-336,-391,-375,-368,-412,-407,-388,-417,-354,-411,-328,-386,-273,-319,-198,-244,-162,-205,-101,-178,-129,-177,-104,-161,-77,-156,-48,-139,-49,-168,-54,-132,-40,-129,-49,-113,-43,-97,-66,-144,-76,-174,-72,-177,-115,-186,-100,-175,-109,-149,-106,-124,-101,-86,-76,-56,-81,-27,-107,3,-87,-27,-99,-52,-77,-57,-80,-68,-65,-62,-77,-17,-71,-16,-67,-12,-47,-11,-33,-14,-33,-31,-4,-26,-4,-79,6,-73,-6,-74,-4,-61,-7,-62,10,-50,33,-27,0,-4,51,21,29,16,43,17,21,38,16,70,-26,47,-23},
+   /* IRC_Composite_C_R0195_T210_P315.wav */
+   {6,-6,6,-6,6,-6,6,-6,6,-6,6,-7,7,-7,7,-7,7,-7,8,-8,8,-8,9,-9,9,-9,9,-9,9,-6,-16,-81,-27,-64,-11,-116,37,-7,308,224,339,624,595,47,-450,456,1108,1630,1528,1730,2147,2381,2098,1287,-731,283,215,213,-551,-518,308,203,729,-194,652,579,759,441,763,571,528,360,442,343,364,229,431,125,293,81,275,88,238,45,215,155,198,82,61,12,5,-51,-58,-119,-156,-200,-208,-255,-244,-261,-244,-232,-236,-252,-249,-226,-258,-256,-283,-248,-281,-296,-308,-266,-304,-256,-333,-220,-281,-198,-262,-186,-220,-164,-184,-148,-198,-168,-234,-150,-270,-190,-317,-221,-358,-246,-358,-268,-356,-277,-357,-276,-375,-272,-357,-257,-384,-239,-370,-230,-374,-258,-374,-262,-363,-281,-344,-263,-317,-224,-292,-194,-243,-172,-216,-149,-196,-138,-151,-112,-138,-121,-140,-121,-158,-136,-149,-135,-124,-140,-124,-136,-125,-160,-126,-165,-139,-151,-144,-151,-125,-132,-116,-99,-80,-92,-84,-64,-97,-68,-84,-76,-83,-93,-75,-65,-93,-87,-89,-70,-103,-55,-102,-68,-112,-68,-70,-61,-55,-62,-34,-41,-24,-58,-48,-63,-63,-52,-80,-56,-81,-54,-68,-55,-48,-48,-24,-69,-21,-60,-6,-21,-10,-12,-6,14,-8,19,1,57,22,38,15,27,32,33},
+   /* IRC_Composite_C_R0195_T225_P315.wav */
+   {-7,7,-8,8,-8,8,-8,8,-8,8,-8,8,-8,8,-8,8,-8,8,-8,9,-9,9,-9,9,-9,9,-9,9,-6,33,28,58,51,52,71,102,56,94,137,218,326,455,483,632,249,92,428,904,1132,1053,1398,1725,1892,1761,996,419,-35,393,230,-426,83,187,561,443,405,624,567,778,322,714,220,555,133,272,57,175,33,-14,10,34,86,73,113,24,-32,-47,-92,-107,-159,-198,-236,-286,-255,-307,-290,-357,-323,-402,-335,-423,-329,-409,-335,-383,-288,-312,-274,-328,-286,-276,-227,-303,-252,-323,-243,-382,-269,-374,-281,-400,-300,-362,-289,-363,-292,-337,-287,-341,-275,-329,-275,-353,-270,-350,-272,-365,-283,-346,-284,-356,-310,-344,-298,-351,-295,-331,-289,-350,-307,-364,-293,-334,-305,-333,-280,-301,-259,-287,-211,-253,-202,-233,-185,-196,-178,-176,-188,-164,-153,-127,-135,-117,-145,-120,-134,-127,-143,-116,-126,-80,-113,-89,-119,-102,-89,-86,-92,-78,-108,-66,-94,-63,-86,-58,-73,-44,-52,-66,-55,-76,-50,-92,-33,-91,-19,-106,-44,-97,-52,-76,-62,-65,-52,-53,-56,-64,-38,-57,-27,-67,-32,-61,-26,-56,-44,-40,-36,-27,-22,-5,-13,-6,12,-15,8,-1,18,1,-3,6,-13,24,6,33,17,44,11,63,31,57,47,73,42,34,14,5},
+   /* IRC_Composite_C_R0195_T240_P315.wav */
+   {0,0,0,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,58,78,16,22,51,114,84,41,96,70,113,111,119,154,190,332,340,437,381,393,351,291,690,869,1203,1249,1386,1308,1507,1378,1185,595,424,365,626,420,458,252,382,637,446,207,255,488,305,321,49,221,53,63,79,-40,108,-82,101,-101,-13,-203,-152,-221,-220,-235,-289,-282,-281,-296,-224,-344,-241,-368,-252,-366,-268,-359,-308,-318,-288,-298,-317,-264,-285,-261,-303,-269,-298,-270,-320,-331,-336,-351,-327,-351,-348,-331,-347,-276,-351,-246,-372,-243,-370,-280,-365,-300,-410,-295,-413,-330,-447,-345,-440,-333,-430,-333,-415,-326,-403,-318,-379,-309,-372,-303,-357,-286,-300,-274,-299,-260,-274,-222,-258,-207,-249,-178,-228,-179,-202,-181,-164,-194,-132,-181,-100,-149,-105,-124,-85,-117,-92,-115,-104,-129,-96,-119,-72,-137,-65,-104,-51,-87,-75,-80,-56,-63,-66,-73,-81,-74,-88,-75,-87,-74,-68,-80,-63,-77,-50,-74,-36,-76,-42,-77,-46,-59,-47,-48,-54,-32,-62,-21,-50,-24,-43,-33,-23,-40,-10,-7,-13,-16,-4,2,7,17,-10,37,-7,35,11,27,-4,32,-1,15,-21,-1,-28,7,-9,16,21,30,38,52,37,48,38,34,3},
+   /* IRC_Composite_C_R0195_T255_P315.wav */
+   {-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-3,3,-3,3,-3,4,-6,74,75,79,61,128,72,79,70,94,189,182,258,204,285,366,406,480,409,503,603,776,875,1073,1366,1703,1601,1585,1344,1253,1054,708,209,184,245,471,220,82,328,234,143,-43,229,63,146,52,46,143,-26,108,-162,-81,-337,-290,-357,-257,-274,-266,-282,-295,-253,-242,-177,-217,-241,-242,-180,-187,-261,-282,-286,-285,-278,-318,-290,-350,-241,-329,-216,-356,-250,-344,-253,-301,-333,-307,-381,-310,-406,-310,-398,-327,-389,-328,-383,-321,-397,-364,-416,-386,-447,-400,-469,-437,-477,-423,-470,-435,-472,-439,-441,-374,-396,-361,-359,-340,-300,-318,-278,-293,-236,-257,-203,-214,-171,-197,-164,-178,-159,-160,-150,-162,-167,-187,-166,-177,-176,-193,-156,-159,-125,-174,-110,-158,-92,-129,-96,-113,-107,-94,-103,-101,-73,-98,-45,-93,-27,-108,-14,-122,-36,-134,-52,-131,-66,-121,-83,-114,-73,-107,-51,-91,-32,-79,2,-74,2,-68,4,-68,20,-45,9,-63,-6,-50,8,-15,-2,-22,-9,6,4,25,-11,21,-12,18,-6,-1,-16,-16,-15,-8,7,8,-10,14,-14,21,-9,18,-34,-11,0,0,38,9,28,24,33,41,32,56,10,45},
+   /* IRC_Composite_C_R0195_T270_P315.wav */
+   {5,-5,5,-5,5,-5,5,-6,6,-6,6,-6,6,-6,7,-7,7,-7,8,-8,8,-9,9,-10,10,-11,12,-13,14,-16,19,-24,35,-112,-87,-172,-147,-119,-127,-152,-113,-103,-89,-113,-68,-116,-46,22,42,151,321,392,548,700,898,1192,1367,1289,1342,1432,1348,1134,1055,823,716,745,988,819,628,418,468,438,382,415,357,462,457,454,331,457,286,349,199,216,110,272,103,141,96,107,157,39,151,90,143,187,135,145,132,119,90,29,67,9,33,-99,-4,-78,-31,-119,-118,-125,-133,-39,-116,6,-53,-9,-2,-28,-8,-68,-109,-175,-231,-243,-320,-296,-388,-312,-439,-323,-431,-308,-403,-341,-379,-320,-333,-304,-317,-322,-321,-302,-340,-277,-367,-233,-348,-216,-344,-197,-304,-195,-255,-199,-226,-173,-179,-167,-168,-175,-177,-169,-197,-168,-238,-156,-231,-167,-240,-168,-219,-192,-218,-202,-205,-205,-186,-197,-157,-172,-143,-163,-152,-143,-150,-149,-177,-148,-159,-149,-150,-163,-154,-183,-160,-189,-132,-190,-124,-156,-96,-125,-92,-99,-99,-74,-85,-61,-83,-67,-68,-74,-62,-91,-54,-91,-46,-108,-22,-86,-15,-70,-6,-63,-3,-43,-35,-76,-52,-91,-99,-89,-83,-72,-67,-70,-42,-53,-31,-59,-32,-49,-39,-55,-65,-43,-43,-36,-50,-25,-39,-34,-26,-31},
+   /* IRC_Composite_C_R0195_T285_P315.wav */
+   {0,0,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,3,-3,0,-53,-54,-56,-85,-47,-114,-135,-181,-153,-193,-267,-244,-55,-96,1,65,-104,-149,-27,77,318,544,620,776,1096,1388,1457,1232,925,941,734,587,918,901,616,762,669,803,637,587,517,641,585,459,366,454,473,472,377,381,369,477,434,303,311,258,424,217,330,203,348,239,206,163,147,190,93,131,3,88,29,61,25,-4,30,-29,-32,-79,-68,-48,-73,-75,-119,-13,-30,17,-48,5,-35,5,-39,-80,-142,-105,-167,-166,-226,-200,-281,-233,-317,-276,-334,-290,-353,-338,-366,-316,-342,-297,-343,-306,-273,-252,-266,-278,-281,-276,-239,-265,-258,-270,-226,-250,-191,-215,-181,-217,-149,-202,-131,-195,-119,-225,-148,-263,-146,-220,-148,-221,-170,-215,-156,-215,-188,-262,-202,-269,-194,-235,-202,-241,-188,-209,-153,-204,-154,-203,-167,-217,-151,-214,-172,-225,-158,-182,-145,-170,-150,-164,-121,-136,-87,-134,-90,-102,-41,-87,-63,-73,-53,-73,-60,-73,-67,-76,-98,-68,-64,-72,-69,-72,-35,-71,-32,-55,-37,-56,-47,-39,-69,-58,-90,-53,-99,-57,-83,-34,-50,-55,-58,-67,-57,-63,-70,-62,-74,-53,-70,-30,-49,-56,-37,-35,-21,-10,-19,-29},
+   /* IRC_Composite_C_R0195_T300_P315.wav */
+   {2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-3,4,-10,-133,-133,-170,-242,-249,-433,-299,-139,-159,192,133,-70,22,-303,18,-19,635,218,706,1507,1430,1383,2387,1796,1075,392,368,909,611,309,-168,353,2,566,420,570,464,600,533,643,640,611,592,536,352,641,435,676,299,466,377,487,301,349,330,262,326,234,228,152,89,163,56,54,-10,7,-78,-45,-66,-85,-68,-104,-18,4,-26,41,2,31,0,79,5,36,26,46,69,42,-19,-67,-82,-80,-133,-166,-222,-198,-222,-228,-274,-302,-279,-261,-261,-282,-253,-280,-237,-241,-240,-230,-249,-229,-231,-221,-230,-219,-245,-228,-234,-188,-202,-201,-222,-167,-182,-137,-203,-165,-172,-150,-180,-163,-208,-169,-194,-164,-229,-193,-245,-204,-252,-223,-246,-236,-257,-266,-272,-274,-271,-280,-267,-251,-243,-225,-207,-200,-174,-184,-165,-180,-160,-164,-160,-158,-167,-156,-169,-151,-138,-125,-125,-108,-104,-80,-94,-71,-82,-75,-85,-57,-68,-57,-49,-58,-52,-65,-51,-59,-53,-74,-44,-57,-44,-43,-49,-55,-42,-57,-55,-65,-60,-62,-53,-54,-78,-70,-64,-55,-66,-73,-66,-48,-68,-87,-49,-57,-32,-77,-35,-66,-33,-71,-35,-70,15,-17,-11,-33},
+   /* IRC_Composite_C_R0195_T315_P315.wav */
+   {-3,3,-4,4,-4,4,-4,4,-4,4,-4,5,-5,5,-5,5,-5,5,-6,6,-6,6,-6,6,-6,6,-5,3,2,-152,-177,-483,-235,-110,-653,-348,-150,-38,208,331,33,-385,-315,-27,364,811,-94,2504,888,2086,3574,3328,531,314,789,729,404,-337,56,36,119,405,418,656,613,503,327,557,477,459,703,343,212,405,546,432,317,280,459,304,364,271,334,99,204,199,155,38,34,17,94,27,75,-28,51,-48,69,-29,44,-56,45,-30,20,5,41,-28,41,31,56,13,68,19,90,-37,31,-65,21,-87,-51,-142,-103,-155,-98,-191,-137,-221,-116,-207,-139,-209,-136,-203,-147,-207,-168,-209,-164,-226,-176,-243,-198,-258,-194,-267,-197,-246,-196,-229,-191,-208,-190,-182,-197,-201,-171,-193,-170,-221,-151,-220,-141,-245,-167,-251,-184,-255,-210,-246,-238,-245,-223,-258,-247,-284,-254,-278,-249,-311,-255,-297,-217,-263,-193,-236,-176,-198,-176,-178,-155,-172,-140,-181,-132,-195,-91,-184,-92,-165,-90,-121,-68,-117,-88,-97,-47,-73,-55,-70,-16,-46,-13,-61,-26,-38,-50,-48,-41,-43,-40,-60,-49,-55,-58,-57,-47,-68,-57,-81,-49,-74,-61,-74,-29,-70,-35,-79,-39,-87,-35,-94,-49,-86,-55,-65,-49,-58,-43,-30,-32,-23,-33,-5,-23,19,-13,-17},
+   /* IRC_Composite_C_R0195_T330_P315.wav */
+   {-7,7,-7,7,-7,7,-7,7,-7,7,-7,7,-7,7,-7,7,-7,8,-8,8,-8,9,-9,9,-10,11,-11,-55,-296,-386,-360,-162,-486,-689,-594,323,-279,489,253,-225,-799,-208,-290,1653,-875,1421,3185,831,3972,4758,3089,-835,674,981,1685,-351,-114,-17,248,390,777,359,517,535,503,290,712,58,681,346,142,-203,198,199,331,-307,34,132,139,70,114,91,13,123,184,191,4,23,205,163,125,72,141,-40,14,124,-28,33,-51,40,-5,35,-58,73,-52,-16,56,78,-40,-37,45,32,27,55,-7,-15,-18,29,-29,22,-117,6,-57,-69,-79,-66,-162,-105,-133,-121,-164,-165,-210,-152,-236,-167,-217,-215,-247,-204,-244,-201,-239,-194,-273,-180,-249,-181,-223,-192,-224,-139,-212,-188,-198,-169,-226,-134,-219,-153,-208,-148,-204,-149,-219,-168,-214,-162,-245,-160,-257,-219,-229,-208,-279,-218,-292,-266,-282,-268,-275,-227,-298,-216,-236,-164,-238,-151,-226,-142,-217,-147,-203,-145,-177,-146,-138,-122,-131,-73,-151,-51,-110,-31,-107,-6,-86,-9,-40,-10,-36,-1,-49,-9,-42,-8,-74,2,-73,3,-80,-26,-79,6,-86,-19,-76,-29,-75,-32,-78,-38,-65,-46,-77,-46,-88,-55,-75,-74,-82,-34,-77,-58,-74,-44,-61,-16,-21,-21,-1,-32,25,-5,-9,-4},
+   /* IRC_Composite_C_R0195_T345_P315.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-2,2,-2,2,-1,1,0,0,1,-2,3,-4,6,-9,13,-23,-205,-456,-570,-178,-609,-516,-1049,-349,515,157,-280,1141,-1675,-135,-1096,1502,703,-567,2409,4580,1615,5569,6660,-925,461,21,3064,-94,275,-939,176,137,1125,718,-145,338,911,169,330,365,552,148,369,-393,179,-127,204,80,-84,-339,8,81,-65,-288,145,50,-22,-24,61,62,45,-71,194,159,54,-19,226,-56,24,48,99,-31,-91,-8,53,24,-79,12,113,-69,-3,43,52,-53,25,22,42,-13,-5,-1,64,-132,-15,-42,-12,-176,-19,-103,-113,-146,-76,-99,-119,-177,-139,-96,-193,-229,-107,-210,-181,-214,-106,-219,-211,-156,-157,-151,-246,-153,-144,-151,-254,-173,-147,-195,-202,-155,-196,-154,-199,-151,-187,-147,-192,-144,-157,-216,-170,-139,-200,-196,-193,-158,-269,-172,-256,-190,-293,-226,-252,-229,-318,-238,-250,-265,-306,-166,-254,-191,-300,-137,-229,-129,-286,-136,-186,-156,-221,-108,-162,-109,-143,-76,-137,-51,-129,-13,-99,-24,-56,17,-62,-20,-41,40,-86,44,-52,39,-123,42,-65,39,-115,-16,-46,0,-101,-28,-69,-10,-115,-27,-101,20,-128,3,-110,13,-133,-3,-101,-28,-103,-3,-79,-10,-82,10,-56,27,-77,25,-50,57,-33,20,-49,39}};
+
+const int16_t irc_composite_c_r0195_p330[][256] =
+  {/* IRC_Composite_C_R0195_T000_P330.wav */
+   {-59,61,-63,65,-67,69,-71,73,-76,78,-81,83,-86,89,-92,95,-98,100,-102,103,-103,99,-90,66,27,-206,-447,-812,-1040,-532,-532,-221,-2282,2737,-1254,634,-40,577,-3131,-820,-289,5598,-4456,4514,11627,-957,5804,7393,108,-5260,1410,3318,1621,-1983,2287,284,1686,1238,2060,-1196,399,489,1551,51,439,-136,620,290,-51,26,548,-418,535,-429,119,-693,256,-317,191,-412,18,-258,-48,-331,42,-245,-151,-84,-244,-168,-112,-248,56,-243,-130,-346,3,-311,-140,-201,-205,-208,-164,-257,-153,-216,-142,-212,-32,-240,-169,-200,-80,-259,-56,-266,-49,-259,-153,-247,-131,-186,-180,-157,-174,-170,-168,-133,-130,-144,-118,-156,-112,-157,-115,-125,-52,-108,-95,-100,-113,-69,-143,-89,-121,-105,-124,-163,-161,-185,-102,-218,-93,-213,-136,-158,-144,-182,-122,-122,-161,-147,-153,-151,-116,-162,-100,-155,-94,-176,-98,-138,-94,-127,-119,-122,-158,-125,-151,-130,-178,-168,-144,-200,-186,-199,-181,-157,-192,-201,-216,-176,-230,-170,-197,-219,-207,-194,-203,-187,-195,-174,-204,-175,-205,-170,-207,-182,-206,-156,-195,-149,-193,-165,-200,-125,-166,-120,-182,-142,-113,-73,-137,-86,-97,-75,-88,-72,-83,-37,-82,-24,-72,-40,-53,24,-38,-23,-48,32,10,7,-19,18,7,43,7,34,32,65,53,56,21,94,55,42},
+   /* IRC_Composite_C_R0195_T015_P330.wav */
+   {-33,32,-32,31,-31,30,-28,27,-25,22,-19,15,-10,4,5,-16,31,-53,84,-136,240,-673,177,-174,-1030,-1321,-76,-1250,-372,289,-2348,2418,1442,-314,-2260,351,-1076,96,-4201,12653,-2793,350,16253,7416,-1734,373,1945,-1722,32,2367,1990,-2906,2200,2573,955,-397,1580,-424,684,371,1473,401,8,-1088,1156,-110,164,-526,361,-233,-337,-706,160,-330,-24,-130,132,-810,170,-491,242,-434,33,-622,355,-505,-308,-68,-61,-264,-71,-244,-294,-284,-179,-295,41,-462,-220,-369,-36,-507,-206,-424,-65,-416,-180,-392,-246,-316,-59,-245,-204,-339,-224,-205,-78,-425,-107,-254,-66,-399,-77,-350,46,-239,-96,-256,-101,-268,-138,-142,-190,-179,-136,-172,-84,-261,-40,-173,25,-283,-32,-282,9,-234,-87,-198,-101,-238,-154,-207,-159,-166,-106,-164,-136,-199,-109,-121,-73,-175,-65,-149,-73,-162,-108,-135,-78,-99,-124,-126,-179,-91,-133,-84,-188,-122,-167,-148,-155,-182,-183,-160,-158,-219,-207,-224,-181,-161,-192,-206,-207,-174,-243,-140,-189,-173,-194,-138,-203,-173,-192,-143,-155,-154,-216,-158,-170,-138,-202,-110,-182,-132,-154,-83,-201,-104,-130,-44,-129,-80,-139,-19,-72,-80,-88,-16,-110,-20,-46,-35,-88,15,-32,6,-67,-9,-16,52,-18,27,27,26,-2,60,51,76,28,98,40,77,33,94,11,62},
+   /* IRC_Composite_C_R0195_T030_P330.wav */
+   {15,-16,18,-19,21,-23,26,-29,32,-36,41,-47,54,-62,73,-86,104,-126,155,-102,601,-1085,-197,-1634,218,-1892,836,-1472,627,332,2248,1679,-4388,-1076,2778,-5228,3611,10385,-5976,10952,11266,9110,-4453,651,-313,-354,33,2898,-224,-1072,2571,1824,-115,539,940,-84,401,1263,337,3,-171,-426,685,-69,-27,-94,-916,-1001,131,-752,-458,-101,-74,-99,-305,-309,-166,-102,238,-512,98,-429,-43,-482,424,-183,-138,-249,-301,-240,-13,-418,-203,-184,-296,-341,-269,-474,-99,-447,-210,-601,-403,-560,-148,-418,-397,-430,-241,-495,-140,-391,-94,-358,-170,-431,-28,-368,-72,-337,-69,-376,-11,-267,-144,-348,-177,-200,-156,-317,-161,-307,-97,-238,-69,-237,-166,-271,-82,-187,-69,-237,-35,-248,-105,-213,-38,-207,-108,-284,-91,-256,-64,-223,-92,-227,-52,-205,-48,-178,-94,-174,-26,-226,-71,-202,-71,-184,-34,-207,-88,-154,-79,-180,-41,-222,-49,-215,-56,-245,-90,-251,-51,-260,-109,-246,-150,-229,-109,-239,-125,-253,-146,-228,-85,-219,-106,-252,-82,-250,-58,-194,-116,-232,-74,-193,-81,-184,-110,-195,-65,-162,-92,-193,-65,-199,-29,-203,0,-193,28,-151,-16,-159,-4,-109,-9,-90,-21,-99,36,-92,27,-82,30,-74,71,-74,64,-43,102,-28,93,18,88,22,98,29,116,21,126,-20,133,-40,143,-66},
+   /* IRC_Composite_C_R0195_T045_P330.wav */
+   {60,-60,60,-60,60,-60,59,-57,55,-51,47,-40,30,-15,-9,50,-138,582,-538,-783,-136,149,-1756,-848,-253,-449,1379,-797,1611,1075,2352,-3846,-4473,1829,1877,4778,13625,-4798,10293,11126,4895,-5087,-2315,-1117,-112,3354,2968,-606,-2642,2599,370,2356,-155,634,-118,50,-430,371,-366,83,83,846,-691,615,-552,-124,-1033,-101,-1482,300,-868,142,-849,-166,-356,234,-620,-26,-586,234,-687,38,-602,184,-537,-8,-502,109,-408,41,-516,144,-611,54,-513,-33,-440,-242,-381,-176,-498,-276,-575,-140,-578,-79,-706,-148,-475,-195,-498,-214,-378,-197,-385,-257,-315,-80,-443,-59,-376,-102,-363,-123,-440,-39,-442,-136,-304,-144,-309,-84,-271,-165,-242,-238,-200,-117,-273,-150,-258,-130,-283,-92,-309,-42,-296,-47,-253,-76,-218,8,-252,-78,-168,-84,-181,-86,-207,-54,-197,-48,-239,-19,-220,-19,-245,-20,-221,-34,-247,-31,-246,-93,-181,-96,-211,-94,-210,-85,-235,-118,-230,-101,-229,-107,-245,-109,-215,-112,-220,-113,-204,-100,-188,-108,-185,-89,-182,-62,-197,-34,-195,-28,-194,-18,-182,-55,-157,-81,-141,-90,-141,-91,-129,-74,-112,-85,-107,-64,-103,-64,-86,-50,-93,-43,-62,-35,-57,-57,-38,-24,-28,-39,-24,-20,-16,22,2,35,17,56,35,75,47,56,59,88,45,61,91,80,67,62,68,59},
+   /* IRC_Composite_C_R0195_T060_P330.wav */
+   {-16,18,-21,24,-27,30,-34,39,-45,51,-59,68,-79,93,-109,120,-488,-209,-843,-446,581,-2538,150,-1048,1674,-298,921,-422,4633,-1379,-3456,-5235,5897,-232,13802,4771,-1871,13790,9150,509,-7321,-2062,426,4376,2825,-66,-4120,2160,1014,1965,-182,10,-66,1012,-1092,605,-694,-103,-630,876,-1446,848,-616,147,-237,208,-605,430,-846,156,-639,-33,-577,-169,-800,-107,-681,-261,-491,-122,-613,-72,-646,-89,-523,-258,-835,-111,-630,-192,-684,-120,-546,-114,-589,-198,-484,-123,-592,-31,-441,-137,-356,-160,-262,-88,-337,-230,-206,-197,-286,-158,-371,-90,-386,-164,-357,-120,-396,-123,-371,-159,-394,-268,-385,-156,-411,-140,-387,-213,-351,-179,-344,-165,-326,-182,-213,-154,-216,-130,-224,-133,-226,-137,-212,-88,-294,-83,-283,-112,-241,-166,-252,-129,-207,-171,-148,-189,-146,-134,-197,-75,-135,-114,-111,-109,-99,-40,-137,-112,-65,-123,-140,-104,-208,-112,-191,-189,-188,-164,-205,-119,-255,-149,-161,-150,-183,-173,-162,-174,-126,-195,-97,-145,-155,-118,-129,-86,-145,-98,-141,-61,-119,-118,-63,-133,-28,-155,2,-161,-3,-128,-39,-100,-78,-65,-82,-72,-98,-47,-85,-70,-46,-75,-12,-101,21,-108,22,-100,40,-92,45,-75,34,-63,31,-40,44,-41,58,-23,74,-14,74,-19,131,-21,113,30,101,49,110,57,108,51},
+   /* IRC_Composite_C_R0195_T075_P330.wav */
+   {19,-20,21,-23,25,-26,28,-31,33,-36,39,-42,45,-49,49,-225,-118,-411,-201,-767,-125,-1198,-1086,1587,699,-114,366,4108,-2207,1045,-7292,822,4055,16836,-2188,1091,15052,6204,1571,-5363,-5126,3112,6714,709,-1466,-3132,1975,682,2154,-1147,580,-537,152,-77,-200,-771,391,-354,-354,-457,389,-921,324,-902,497,-455,-10,-125,532,-709,91,-249,-171,-384,-149,-591,-267,-649,-63,-714,-477,-400,-204,-742,-437,-554,-445,-510,-517,-561,-379,-611,-347,-592,-417,-611,-301,-482,-385,-503,-245,-302,-310,-258,-221,-225,-194,-239,-127,-248,-118,-185,-116,-277,-78,-132,-151,-167,-94,-212,-171,-287,-162,-301,-224,-259,-202,-317,-238,-263,-235,-226,-328,-285,-227,-274,-225,-257,-270,-272,-203,-280,-198,-229,-168,-195,-197,-135,-134,-188,-159,-143,-143,-184,-122,-175,-115,-188,-85,-144,-140,-133,-147,-139,-167,-127,-183,-140,-187,-127,-176,-160,-172,-135,-190,-179,-159,-144,-201,-142,-175,-140,-206,-117,-148,-146,-179,-121,-175,-129,-135,-92,-156,-126,-159,-68,-176,-108,-129,-91,-133,-75,-91,-67,-116,-63,-52,-31,-122,-11,-115,3,-129,15,-110,-22,-119,-11,-83,-62,-34,-39,-48,-27,-43,-19,-79,23,-48,-1,-86,40,-64,16,-37,23,-6,-6,9,18,18,23,24,45,17,82,40,87,28,105,40,116,77,81,80,85},
+   /* IRC_Composite_C_R0195_T090_P330.wav */
+   {6,-4,3,-1,0,3,-5,8,-11,16,-21,29,-39,57,-101,-332,89,-835,-16,8,-1022,-629,-474,1050,924,-984,2683,3640,-2207,-5863,2261,-3476,15168,2953,-3266,10727,14390,3174,-8455,-1966,1449,7129,3075,-3700,-2692,1997,423,1234,-1087,216,-471,1061,-1107,458,-1402,142,319,-88,-425,76,-1043,226,-230,12,-536,-207,-663,129,-455,253,-222,-284,-472,86,-636,265,-368,-261,-539,-160,-407,43,-778,-256,-683,-475,-586,-306,-628,-455,-647,-525,-513,-416,-500,-430,-812,-419,-489,-274,-452,-386,-597,-258,-363,-159,-236,-327,-241,-259,-214,-139,-145,-206,-216,-123,-79,-29,-96,-65,-217,-263,-177,-100,-202,-186,-272,-223,-240,-138,-180,-221,-352,-224,-206,-195,-213,-254,-233,-244,-227,-156,-224,-178,-249,-190,-199,-133,-154,-164,-183,-191,-143,-185,-101,-190,-166,-201,-98,-150,-133,-119,-165,-142,-178,-120,-115,-180,-149,-197,-152,-187,-75,-218,-143,-264,-128,-205,-139,-221,-148,-269,-159,-215,-126,-214,-123,-256,-105,-233,-71,-203,-97,-215,-79,-182,-53,-120,-82,-122,-49,-119,20,-120,24,-141,6,-133,24,-115,14,-116,-39,-71,-8,-71,6,-93,-11,-108,40,-59,43,-147,22,-98,28,-64,17,-87,-13,-67,63,-36,44,-53,62,-77,123,-52,145,-81,82,-33,141,-4,116,8,98,14,137,56,139,-7,138,4},
+   /* IRC_Composite_C_R0195_T105_P330.wav */
+   {61,-61,61,-61,62,-62,62,-62,63,-63,63,-64,64,-65,67,-301,294,-741,-246,-201,457,-1576,-144,-73,2001,-1275,1763,1153,6179,-10039,184,1999,2771,10917,-378,522,13686,11704,-7376,-5609,1748,7382,4547,-1426,-5565,-24,2068,630,-441,-1191,186,450,-243,220,-909,-111,-78,770,-727,124,-899,-156,-79,202,-685,-336,-650,-314,-474,-53,-350,-261,-756,191,-463,128,-417,-139,-484,-53,-226,-113,-375,-280,-491,-342,-457,-195,-589,-428,-623,-328,-572,-393,-551,-458,-577,-399,-453,-401,-526,-392,-496,-278,-442,-319,-518,-259,-377,-158,-346,-171,-366,-187,-308,-53,-156,-44,-233,-111,-203,-9,-156,-124,-286,-249,-248,-172,-147,-210,-273,-296,-215,-151,-154,-166,-209,-170,-207,-76,-118,-174,-253,-208,-154,-172,-145,-172,-242,-211,-169,-70,-176,-103,-198,-93,-133,-36,-100,-118,-191,-149,-134,-127,-138,-220,-229,-254,-179,-191,-192,-230,-211,-235,-161,-158,-161,-230,-195,-206,-174,-165,-144,-216,-190,-173,-127,-136,-156,-157,-179,-150,-128,-122,-138,-124,-152,-83,-109,-30,-73,-66,-103,-55,-77,-32,-75,-113,-125,-86,-47,-60,-39,-89,-29,-60,20,-18,-21,-32,-8,-50,24,-43,-1,-69,-6,-57,11,-53,21,-58,-2,-13,35,-10,30,5,44,12,60,31,79,-4,79,46,97,47,97,36,128,43,132,30,81,38},
+   /* IRC_Composite_C_R0195_T120_P330.wav */
+   {1,4,-8,14,-20,27,-35,44,-54,66,-79,95,-114,135,-159,177,-87,615,-830,163,-909,1304,-1626,-325,297,1206,69,-817,4405,1017,-2560,-616,-3770,4387,14092,-5772,3858,10124,9882,-2074,-2959,-3337,7410,7437,-1780,-6636,-2212,2639,1700,-1078,-1371,-262,1153,-566,417,-320,27,-39,-111,485,-441,-792,-288,-146,-108,-616,-333,-464,-419,-367,-303,-262,-308,-516,-133,-270,-200,-382,-324,-225,-347,-205,-250,-254,-201,-400,-249,-330,-242,-364,-366,-273,-444,-377,-352,-336,-480,-405,-375,-496,-436,-362,-461,-436,-395,-354,-445,-362,-274,-397,-353,-222,-328,-290,-351,-239,-329,-117,-69,-109,-182,-35,-73,-135,-239,-228,-363,-234,-203,-243,-333,-218,-233,-231,-229,-116,-190,-172,-177,-158,-144,-144,-172,-173,-215,-124,-180,-106,-203,-150,-185,-121,-123,-149,-106,-93,-132,-84,-116,-33,-138,-72,-186,-81,-187,-113,-205,-194,-255,-235,-238,-195,-263,-202,-264,-198,-268,-158,-241,-175,-267,-183,-249,-160,-199,-167,-203,-153,-184,-118,-137,-101,-164,-93,-143,-66,-128,-30,-127,-42,-113,-31,-78,-23,-59,-78,-78,-48,-79,-53,-70,-40,-106,-22,-46,-9,-51,-12,-37,-31,-54,-23,-49,-43,-68,-43,-55,-39,-14,-29,-16,-13,-8,32,0,28,5,31,33,68,33,69,42,71,38,89,64,82,64,81,94,57,103,62,51,87},
+   /* IRC_Composite_C_R0195_T135_P330.wav */
+   {13,-11,8,-5,2,2,-7,12,-18,26,-34,45,-58,75,-96,126,-173,263,-756,239,-382,327,-504,374,-988,304,-762,1963,32,1243,612,772,3408,-6680,-2100,10509,4890,-1044,6633,2426,10000,1888,-3554,-3952,7154,5846,-3202,-5792,-2133,1379,2382,-1044,-683,-846,1056,-24,511,18,303,-710,-25,-39,-85,-861,-206,-321,-34,-428,-245,-428,-30,-639,-166,-459,-198,-463,-354,-339,-255,-329,-348,-224,-286,-255,-287,-204,-181,-232,-210,-296,-199,-313,-262,-347,-268,-316,-307,-365,-310,-360,-308,-316,-324,-335,-371,-355,-377,-361,-310,-401,-283,-402,-244,-368,-238,-331,-274,-247,-165,-162,-178,-185,-197,-205,-193,-250,-203,-288,-187,-284,-187,-262,-183,-219,-206,-232,-196,-178,-177,-216,-242,-166,-214,-147,-219,-161,-195,-148,-178,-168,-190,-170,-141,-148,-164,-119,-116,-35,-121,-54,-95,-56,-120,-107,-112,-149,-152,-172,-185,-195,-227,-175,-253,-211,-280,-223,-250,-252,-243,-251,-235,-239,-213,-184,-218,-176,-225,-139,-205,-123,-184,-114,-154,-113,-98,-100,-89,-89,-91,-67,-61,-32,-65,-41,-68,-21,-70,-18,-87,-13,-97,-10,-79,-5,-41,-42,-23,-67,8,-74,15,-75,-43,-63,-41,-18,-81,-7,-69,6,-49,2,-9,-7,21,-8,24,-11,44,32,44,38,36,66,51,91,52,103,44,90,55,87,41,60,28,49},
+   /* IRC_Composite_C_R0195_T150_P330.wav */
+   {15,-18,20,-22,25,-28,31,-35,38,-42,47,-52,57,-63,70,-76,84,-91,97,-95,51,-395,348,-546,384,-515,230,-347,78,797,314,1274,475,2177,1227,-4216,-2348,10076,258,3084,3609,6149,5472,3566,-4049,-1019,5212,3739,-5321,-3637,-1580,2270,1066,-633,-1395,-136,1340,-32,75,-243,507,-628,158,108,-226,-702,-291,-112,-126,-404,-330,-254,-212,-362,-151,-436,-14,-424,-215,-583,-44,-508,-144,-379,-62,-360,-158,-295,-27,-305,-69,-339,-66,-367,-167,-326,-112,-361,-177,-468,-151,-402,-103,-407,-145,-397,-208,-387,-142,-366,-207,-368,-232,-411,-224,-349,-159,-358,-152,-366,-101,-313,-105,-307,-196,-327,-268,-341,-250,-319,-245,-350,-228,-325,-151,-267,-108,-294,-109,-251,-84,-251,-104,-207,-91,-253,-103,-244,-91,-252,-85,-272,-130,-311,-129,-241,-94,-173,-90,-147,-80,-140,-35,-126,-79,-186,-118,-197,-120,-210,-155,-272,-176,-269,-156,-268,-151,-277,-170,-275,-153,-283,-181,-265,-186,-257,-194,-217,-154,-209,-128,-189,-106,-201,-80,-156,-34,-154,-17,-151,12,-97,27,-70,9,-39,-39,-3,-46,-29,-50,-61,-28,-93,-9,-89,-8,-103,1,-74,-1,-70,-9,-35,-44,-51,-51,-29,-31,-30,-13,-35,10,8,7,-4,11,13,35,12,38,37,49,36,56,47,69,81,52,69,15,90,11,52,-17,56},
+   /* IRC_Composite_C_R0195_T165_P330.wav */
+   {-38,38,-39,39,-39,40,-40,41,-42,43,-44,46,-47,49,-52,55,-59,64,-70,79,-90,108,-143,381,-197,374,-312,378,-424,566,-428,1134,331,1309,501,1274,2077,-920,-3821,4175,5452,953,5098,2533,5399,3026,943,-4554,1848,1742,-912,-4176,-875,-267,1882,-418,-1283,-863,1021,217,-28,-198,445,-494,-277,-183,13,-551,-347,-206,-115,-218,-241,-334,-209,-134,-196,-318,-365,-189,-242,-376,-315,-159,-285,-247,-217,-119,-295,-192,-233,-63,-298,-72,-221,-78,-309,-142,-336,-207,-384,-220,-389,-223,-371,-217,-392,-165,-378,-175,-375,-156,-391,-200,-346,-196,-386,-169,-293,-171,-325,-127,-312,-180,-413,-244,-413,-275,-437,-268,-434,-258,-381,-194,-377,-183,-352,-126,-339,-126,-286,-92,-262,-50,-229,-59,-226,-70,-231,-92,-226,-106,-251,-108,-219,-79,-217,-73,-199,-48,-168,-54,-223,-113,-224,-121,-228,-134,-243,-180,-267,-148,-265,-176,-290,-166,-274,-160,-279,-151,-277,-159,-262,-146,-233,-125,-202,-129,-198,-90,-162,-74,-165,-69,-153,-33,-141,-13,-136,2,-114,40,-123,47,-102,43,-95,48,-108,50,-112,44,-100,32,-92,18,-95,22,-82,17,-82,23,-102,19,-84,22,-128,24,-87,44,-69,35,-66,63,-29,79,-26,54,-19,109,-9,110,-5,137,-8,111,4,120,34,92,-13,96,-16,93,-54,92},
+   /* IRC_Composite_C_R0195_T180_P330.wav */
+   {-26,26,-26,25,-25,25,-24,23,-23,22,-21,20,-19,17,-16,14,-11,8,-4,-1,7,-15,27,-46,89,61,-145,147,-71,284,-270,217,-379,526,177,1579,111,1304,-624,3003,-1570,-1854,2371,5215,2766,3717,2594,2071,4289,53,-1269,-2397,2282,-1207,-426,-1536,-96,41,749,-948,-416,181,838,-208,144,-15,-109,-323,76,15,16,-276,4,-8,-1,-123,-108,-209,-91,-130,71,-58,-162,-230,-119,-177,-166,-207,-161,-106,-120,-36,-71,-88,-115,-141,-23,-102,-161,-225,-242,-194,-306,-250,-347,-260,-382,-276,-340,-235,-319,-231,-283,-196,-267,-199,-293,-242,-296,-261,-227,-164,-163,-202,-268,-256,-325,-304,-440,-389,-451,-306,-367,-292,-374,-282,-281,-248,-233,-251,-211,-225,-190,-161,-223,-149,-241,-144,-229,-124,-173,-155,-158,-194,-95,-175,-83,-151,-123,-82,-138,-35,-195,-70,-203,-140,-178,-222,-181,-287,-200,-285,-259,-281,-289,-246,-302,-246,-260,-227,-227,-242,-189,-224,-180,-220,-169,-184,-146,-121,-121,-86,-121,-59,-86,-47,-78,-63,-71,-90,-66,-82,-53,-82,-82,-54,-72,-16,-90,-24,-88,-44,-31,-45,-32,-70,-43,-25,-59,-33,-76,-24,-59,-46,-55,-45,-56,-56,-64,-32,-54,-6,-57,-1,-22,23,2,9,-2,35,54,34,72,22,84,15,94,10,75,34,67,47,21,57,14,36},
+   /* IRC_Composite_C_R0195_T195_P330.wav */
+   {19,-19,19,-20,20,-21,21,-22,23,-23,24,-24,25,-26,26,-27,27,-28,29,-29,30,-30,30,-29,28,-25,18,14,179,64,-45,11,282,14,-136,-276,1166,711,738,-60,1508,162,166,165,449,1532,5355,3392,931,2171,2302,2446,-1456,-644,-2020,1267,216,-603,-1662,-126,427,-150,-158,310,15,459,-99,240,14,137,-140,123,54,92,-77,39,37,33,-11,20,88,136,-9,51,-50,-40,-148,-156,-244,-108,-193,-138,-117,-78,-113,-124,-103,-88,-193,-190,-214,-191,-267,-312,-314,-321,-363,-318,-371,-313,-329,-304,-279,-293,-244,-274,-270,-284,-268,-270,-226,-229,-218,-220,-244,-283,-323,-321,-343,-387,-412,-395,-369,-338,-385,-339,-361,-243,-279,-233,-263,-196,-226,-166,-214,-179,-210,-188,-205,-188,-194,-186,-176,-189,-187,-188,-151,-148,-132,-148,-138,-144,-161,-155,-165,-177,-165,-222,-217,-269,-237,-294,-251,-318,-235,-297,-234,-275,-203,-229,-195,-221,-176,-219,-151,-204,-120,-189,-136,-166,-97,-148,-79,-121,-68,-94,-95,-78,-72,-67,-80,-75,-79,-69,-70,-61,-76,-83,-71,-84,-52,-102,-40,-87,-65,-74,-59,-59,-44,-53,-35,-32,-38,-24,-30,-33,-41,-31,-32,-32,-53,-10,-53,7,-39,3,11,-3,30,29,44,32,63,46,79,39,81,48,51,38,73,36,50,22,77,-11},
+   /* IRC_Composite_C_R0195_T210_P330.wav */
+   {4,-4,4,-4,4,-4,4,-4,4,-4,4,-4,4,-4,4,-4,4,-4,4,-3,3,-3,2,-2,1,0,-1,4,-7,14,-37,-66,67,-14,-33,-186,91,-96,90,124,958,676,187,107,-283,310,1299,1265,1017,1358,3762,3924,898,885,85,1188,235,180,-1266,-91,662,-183,-222,-323,535,278,681,120,382,366,409,299,413,216,314,49,299,205,244,15,349,134,369,-25,355,37,155,-122,149,-90,17,-126,0,-119,-101,-98,-45,-101,-108,-137,-41,-165,-128,-190,-90,-220,-248,-213,-218,-183,-289,-234,-287,-213,-287,-181,-291,-252,-334,-216,-249,-215,-305,-231,-259,-234,-279,-267,-321,-303,-381,-272,-334,-278,-372,-241,-327,-217,-356,-191,-320,-190,-311,-163,-271,-146,-273,-141,-249,-181,-245,-176,-249,-177,-262,-171,-258,-201,-269,-174,-249,-165,-249,-157,-236,-157,-222,-150,-201,-185,-232,-186,-231,-212,-272,-199,-268,-197,-269,-185,-248,-203,-236,-188,-226,-202,-206,-202,-200,-198,-175,-174,-139,-154,-129,-144,-119,-116,-115,-116,-101,-118,-88,-110,-115,-124,-128,-104,-139,-117,-161,-110,-128,-96,-116,-100,-91,-78,-46,-54,-38,-54,-28,-28,-31,-22,-47,-22,-45,-45,-34,-47,-26,-63,-19,-69,15,-46,17,-34,36,-7,28,11,12,21,18,27,33,12,47,14,27,2,38,-2},
+   /* IRC_Composite_C_R0195_T225_P330.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,2,-68,-59,-5,-96,48,-93,-175,-114,61,113,32,173,565,369,-5,6,42,629,908,1232,1079,1509,2842,2279,709,505,658,652,456,31,-132,16,714,22,392,243,724,581,610,769,414,780,295,589,257,440,139,342,210,326,138,213,205,269,65,206,39,192,-75,77,-72,119,-93,32,-98,-25,-124,-61,-115,-124,-100,-152,-32,-169,-66,-171,-60,-174,-142,-204,-153,-209,-250,-258,-282,-284,-268,-319,-270,-333,-243,-293,-243,-293,-258,-255,-244,-253,-268,-246,-247,-251,-255,-254,-239,-252,-237,-244,-241,-235,-232,-243,-208,-257,-193,-268,-204,-268,-227,-248,-235,-265,-257,-230,-233,-218,-240,-222,-215,-193,-210,-173,-175,-183,-176,-191,-180,-202,-196,-223,-223,-252,-229,-258,-254,-257,-250,-252,-220,-262,-227,-250,-215,-243,-223,-230,-208,-202,-158,-170,-134,-160,-123,-166,-123,-164,-100,-174,-109,-180,-112,-164,-109,-168,-131,-171,-106,-158,-104,-162,-121,-153,-114,-132,-112,-125,-95,-82,-74,-52,-73,-50,-54,-44,-32,-38,-20,-57,-17,-36,-14,-46,-36,-11,-10,-23,-34,-7,-16,7,-20,-30,-21,-13,-19,-3,-32,-10,-25,9,-16,17,18,-2,23},
+   /* IRC_Composite_C_R0195_T240_P330.wav */
+   {6,-6,6,-6,6,-6,6,-6,7,-7,7,-7,7,-7,7,-7,7,-7,7,-7,8,-8,8,-8,8,-8,8,-8,1,-47,-37,-124,-93,-95,-58,-38,-70,-58,-105,-121,-151,-76,14,175,104,51,147,291,191,-40,158,429,1090,1311,1373,1241,1326,1526,1181,419,583,828,657,621,656,466,551,572,739,747,685,612,423,667,583,601,619,480,483,323,428,299,276,192,162,105,83,89,89,51,98,-46,97,-2,117,-65,34,-69,1,-69,-69,-127,-170,-131,-173,-121,-145,-147,-134,-204,-102,-232,-83,-273,-123,-325,-159,-309,-183,-275,-249,-249,-252,-185,-266,-178,-245,-201,-259,-208,-248,-243,-270,-238,-253,-227,-254,-209,-249,-203,-269,-184,-271,-141,-258,-121,-230,-113,-188,-120,-198,-167,-178,-200,-214,-242,-233,-264,-244,-276,-244,-267,-267,-261,-267,-239,-274,-233,-272,-226,-277,-260,-268,-247,-282,-253,-289,-226,-270,-213,-255,-189,-247,-158,-192,-146,-173,-170,-171,-177,-143,-184,-151,-187,-146,-147,-141,-154,-121,-150,-139,-166,-139,-155,-135,-176,-141,-174,-144,-174,-160,-177,-162,-156,-152,-157,-114,-126,-89,-103,-86,-98,-78,-68,-62,-53,-84,-35,-53,-7,-51,-15,-38,-6,-34,-13,-34,-15,-32,-28,-35,-19,-32,1,-19,7,-27,9,3,4,-14,3,-5,36,2,41},
+   /* IRC_Composite_C_R0195_T255_P330.wav */
+   {-1,1,-1,1,-1,1,-2,2,-2,2,-2,2,-2,2,-2,2,-2,3,-3,3,-3,3,-3,4,-4,4,-5,5,-6,7,-8,10,-13,22,-83,-97,-106,-152,-99,-59,-90,-92,-128,-74,-117,-122,-12,68,86,137,156,199,190,329,533,767,1036,1233,1574,1651,1491,1169,954,781,906,1168,950,370,704,786,629,605,494,653,561,698,550,738,501,619,385,441,218,218,189,132,100,39,147,27,91,-38,77,-24,27,-36,-30,-17,-41,-95,-49,-122,-68,-169,-78,-186,-103,-225,-135,-197,-120,-153,-145,-183,-142,-167,-150,-231,-159,-244,-153,-214,-165,-219,-193,-224,-173,-215,-157,-238,-162,-231,-149,-238,-155,-243,-185,-203,-182,-175,-206,-184,-214,-165,-184,-162,-168,-165,-138,-160,-133,-192,-158,-236,-216,-272,-273,-290,-295,-306,-316,-279,-301,-261,-303,-259,-299,-287,-289,-292,-273,-302,-247,-283,-207,-253,-202,-230,-201,-189,-205,-172,-192,-162,-193,-165,-169,-166,-191,-174,-179,-167,-155,-172,-157,-151,-142,-148,-140,-156,-161,-162,-168,-167,-172,-156,-177,-164,-174,-152,-168,-156,-171,-153,-173,-152,-171,-139,-162,-124,-144,-85,-111,-53,-108,-40,-71,-33,-75,-21,-62,-20,-68,0,-50,5,-37,-3,-30,28,-23,3,-23,12,-19,28,-12,24,-5,41,17,24,-7,17,-24,23},
+   /* IRC_Composite_C_R0195_T270_P330.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,1,-2,3,-127,-152,-188,-165,-160,-178,-195,-131,-72,-37,-53,-57,-34,48,106,135,127,188,385,765,884,1201,1389,1575,1756,1541,1264,1282,908,615,982,1053,498,403,667,706,647,681,663,732,712,700,422,527,505,426,401,192,156,184,190,100,26,-34,55,-50,19,-119,28,-59,-54,-94,-103,-24,-139,-24,-165,-13,-154,-5,-159,-81,-104,-118,-99,-136,-98,-118,-106,-89,-142,-101,-168,-112,-178,-178,-148,-180,-114,-217,-134,-255,-104,-252,-140,-238,-148,-235,-187,-236,-202,-242,-197,-183,-140,-137,-108,-143,-129,-142,-128,-211,-166,-246,-182,-269,-212,-271,-233,-292,-267,-287,-263,-282,-269,-312,-249,-304,-257,-293,-240,-287,-230,-257,-242,-248,-234,-224,-236,-219,-241,-188,-227,-176,-221,-206,-210,-185,-185,-166,-179,-163,-176,-164,-168,-164,-186,-168,-180,-154,-159,-141,-142,-150,-142,-147,-152,-145,-159,-150,-168,-167,-187,-177,-186,-176,-192,-163,-197,-167,-186,-164,-170,-159,-148,-137,-112,-102,-104,-88,-70,-56,-50,-28,-30,-11,-12,-16,-10,-21,-31,-4,-26,2,15,8,10,-3,-2,5,10,12,-9,0,-5,9,-13,-19,-29,-16},
+   /* IRC_Composite_C_R0195_T285_P330.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,1,-1,2,-2,3,-6,13,-111,-184,-215,-184,-140,-139,-179,-220,-168,-247,-124,1,-91,-84,42,-80,17,-54,56,218,634,881,967,1181,1587,1636,1273,1643,1650,702,357,915,823,759,711,555,754,615,798,685,720,707,793,638,630,510,595,460,429,211,188,178,148,102,16,162,0,142,-40,140,-32,103,24,-7,-30,11,-24,-1,-87,12,-168,51,-103,64,-105,39,-118,10,-91,-91,-119,-125,-90,-149,-75,-130,-138,-153,-195,-185,-254,-205,-313,-223,-303,-243,-266,-224,-223,-230,-188,-216,-155,-167,-129,-140,-137,-102,-110,-74,-146,-82,-177,-89,-192,-110,-215,-160,-231,-232,-220,-242,-237,-262,-238,-271,-237,-259,-263,-234,-261,-228,-278,-222,-281,-252,-268,-270,-265,-263,-228,-244,-187,-230,-176,-214,-164,-194,-141,-175,-152,-173,-143,-161,-171,-179,-174,-184,-157,-194,-163,-211,-148,-196,-144,-205,-158,-219,-155,-190,-162,-178,-166,-174,-156,-186,-169,-200,-161,-200,-167,-208,-167,-218,-158,-186,-143,-155,-119,-123,-104,-117,-86,-101,-54,-82,-39,-73,-25,-68,-7,-55,-18,-42,23,-14,19,-21,7,-24,19,-4,2,-14,7,-10,12,-14,13,-7,26,-23,8,-14},
+   /* IRC_Composite_C_R0195_T300_P330.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,4,-4,4,-4,4,-4,4,-5,-62,-157,-83,-247,-284,-202,-303,-294,-235,-54,-305,15,156,241,-112,-479,-226,774,-232,495,1765,1236,1265,2221,2705,1763,850,505,963,731,757,546,519,206,795,441,708,577,742,596,545,491,526,666,413,377,307,300,457,188,260,34,250,133,251,58,87,249,132,194,-1,200,55,156,95,139,122,50,62,57,84,-3,-71,-62,-99,-113,-176,-122,-254,-180,-227,-166,-230,-202,-293,-196,-279,-248,-326,-256,-320,-217,-260,-211,-244,-154,-191,-125,-171,-113,-160,-81,-135,-55,-102,-70,-95,-24,-42,-53,-78,-73,-93,-89,-149,-139,-195,-180,-235,-192,-245,-233,-268,-234,-245,-224,-256,-221,-249,-225,-220,-233,-215,-250,-226,-218,-236,-221,-224,-181,-198,-176,-191,-173,-185,-156,-184,-162,-178,-171,-172,-156,-184,-183,-174,-192,-186,-179,-189,-180,-212,-184,-191,-167,-199,-217,-182,-203,-172,-210,-178,-224,-155,-220,-185,-224,-199,-194,-203,-201,-206,-192,-196,-188,-165,-158,-177,-147,-145,-112,-128,-98,-113,-74,-77,-54,-65,-67,-32,-44,-10,-54,-13,-29,-7,-15,-1,-23,19,-18,10,-27,11,-16,8,2,12,-12,4,-1,37,18,19,17,36},
+   /* IRC_Composite_C_R0195_T315_P330.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-3,-227,-204,-310,-344,-179,-311,-505,-298,-88,-402,115,712,-76,-577,-211,9,558,-404,914,2627,1047,2118,4245,2542,908,788,1382,865,422,674,-11,248,192,1314,191,817,366,966,306,467,192,289,334,100,222,7,227,236,315,254,209,305,479,300,240,335,445,260,199,112,202,84,42,63,85,-45,-100,-80,-73,-157,-181,-120,-209,-167,-259,-161,-262,-234,-281,-209,-243,-208,-221,-245,-173,-195,-197,-241,-151,-201,-166,-240,-132,-182,-104,-150,-85,-104,-75,-148,-59,-141,-36,-144,-33,-133,-71,-89,-7,-47,-109,-70,-108,-44,-152,-113,-164,-122,-182,-216,-170,-212,-172,-263,-155,-228,-178,-252,-176,-207,-200,-242,-206,-220,-204,-208,-195,-218,-188,-193,-156,-217,-159,-200,-119,-211,-133,-181,-131,-177,-169,-158,-166,-171,-195,-175,-181,-214,-193,-224,-184,-218,-204,-219,-183,-222,-197,-222,-197,-226,-215,-204,-205,-207,-216,-212,-210,-224,-222,-213,-210,-205,-202,-194,-174,-155,-148,-134,-144,-126,-89,-101,-75,-94,-84,-69,-47,-37,-69,-67,-26,-35,-13,-29,-17,-23,-3,13,-18,-8,-26,22,-30,20,-15,30,-13,8,-11,23,49,30,39,22,36,16},
+   /* IRC_Composite_C_R0195_T330_P330.wav */
+   {9,-9,9,-10,10,-10,10,-11,11,-12,12,-12,13,-13,14,-15,15,-16,17,-17,18,-19,20,-22,23,-25,30,-50,-332,-388,-131,-617,-214,-313,-733,-659,255,-155,-66,699,-77,-372,-1255,-242,2041,-707,-6,4736,2036,2809,3545,4509,22,-72,1207,1973,-173,508,959,302,542,540,656,123,461,665,400,314,251,504,190,265,35,373,467,91,154,128,447,203,170,90,232,80,149,193,21,-6,-90,-101,-59,-80,-255,-144,-154,-358,-29,-238,-96,-237,-126,-195,15,-288,-107,-147,-89,-276,-120,-223,-134,-174,-181,-152,-158,-179,-71,-138,-116,-177,-93,-189,-60,-206,-162,-159,-113,-111,-150,-123,-93,-106,-83,-101,-52,-113,-43,-99,-10,-99,-66,-88,-58,-127,-64,-152,-99,-152,-144,-200,-127,-259,-116,-227,-129,-245,-98,-258,-94,-239,-152,-227,-154,-225,-147,-242,-169,-202,-172,-212,-140,-232,-141,-182,-139,-168,-130,-205,-127,-137,-140,-176,-175,-160,-166,-171,-205,-188,-193,-212,-199,-238,-185,-248,-173,-246,-186,-227,-206,-245,-184,-218,-210,-235,-210,-228,-175,-238,-206,-213,-189,-196,-171,-200,-159,-123,-146,-114,-128,-128,-81,-88,-83,-80,-52,-124,-70,-58,-40,-45,-69,-50,-10,-13,-16,-42,-4,-25,50,-37,11,-37,37,-4,24,-11,12,-7,33,10,69,29,60,20,87,-1},
+   /* IRC_Composite_C_R0195_T345_P330.wav */
+   {1,0,0,0,-1,1,-2,2,-3,4,-5,6,-7,8,-9,11,-13,15,-17,20,-24,28,-33,40,-47,52,-211,-235,-688,-550,-297,-690,-704,-444,713,-1568,989,470,-39,-1205,323,-714,276,-2380,5748,2929,602,8080,5302,-453,-722,3812,715,-74,-365,1626,369,1163,1544,781,-4,229,668,441,364,676,214,441,36,409,174,565,-210,423,-90,172,-132,46,-99,78,-186,159,-181,96,-244,13,-280,-20,-156,-41,-291,-109,-365,-55,-322,-117,-274,-136,-169,-159,-189,-140,-191,12,-178,0,-295,-70,-206,-88,-236,-75,-169,-78,-191,-95,-204,-87,-146,-66,-185,-107,-238,-113,-249,-119,-185,-98,-187,-131,-176,-95,-137,-31,-142,-17,-163,-50,-120,-40,-114,-19,-100,-43,-142,-64,-163,-81,-167,-107,-184,-140,-220,-122,-200,-146,-203,-155,-165,-178,-177,-198,-166,-175,-153,-160,-161,-178,-163,-171,-143,-138,-136,-139,-146,-131,-151,-132,-167,-143,-153,-146,-165,-157,-219,-159,-215,-151,-222,-190,-225,-204,-218,-208,-214,-189,-218,-191,-221,-190,-214,-185,-216,-196,-215,-178,-200,-217,-218,-194,-162,-182,-170,-189,-185,-152,-157,-123,-144,-122,-111,-99,-99,-89,-76,-63,-81,-56,-61,-31,-49,-85,-23,-30,-14,-9,-21,-45,-18,-1,16,3,5,-5,5,8,-5,64,39,55,37,73,42,56,31,84}};
+
+const int16_t irc_composite_c_r0195_p345[][256] =
+  {/* IRC_Composite_C_R0195_T000_P345.wav */
+   {2,-1,1,-1,1,-1,1,-1,0,0,0,0,-1,1,-2,2,-3,4,-5,7,-9,11,-15,21,-203,-88,-927,-196,-604,-22,-2141,-496,-650,1585,-548,803,-263,412,-5131,5346,-2103,-7267,12288,1672,1403,10338,7456,-3182,-874,2680,1965,-5296,1356,1865,68,-578,2523,1092,1554,2056,2530,1058,1432,379,1062,-82,-380,-971,369,-376,62,-56,669,-390,190,379,537,98,-46,-35,315,348,110,183,491,-38,19,211,170,-22,-231,-125,-221,138,-512,-336,-126,10,-387,-311,-283,-176,-241,-345,-432,-207,-330,-489,-386,-392,-351,-423,-274,-457,-352,-361,-311,-213,-391,-323,-302,-143,-325,-378,-206,-229,-190,-319,-334,-196,-236,-126,-387,-117,-339,-105,-264,-135,-207,-172,-176,-254,-160,-183,-177,-200,-218,-201,-189,-116,-205,-120,-224,-113,-159,-119,-241,-118,-127,-119,-188,-91,-119,-15,-186,-30,-132,28,-213,-19,-166,21,-190,-46,-185,-49,-159,-65,-168,-65,-190,-59,-218,-46,-217,-24,-231,-74,-204,-52,-181,-89,-211,-94,-196,-70,-219,-104,-200,-55,-181,-86,-186,-39,-155,-66,-178,-1,-187,-47,-188,6,-142,-11,-161,-5,-114,-9,-113,6,-145,-7,-123,-21,-161,-34,-116,-47,-173,-82,-145,-76,-157,-72,-137,-75,-186,-79,-163,-31,-131,-17,-156,-51,-99,26,-74,-5,-101,37,-82,48,-83,35,-130,20},
+   /* IRC_Composite_C_R0195_T015_P345.wav */
+   {-8,8,-9,9,-9,10,-10,11,-11,12,-13,14,-15,17,-19,21,-23,26,-30,34,-31,-453,-1036,-33,-665,328,-1202,-749,-1695,-16,154,1160,807,430,-250,202,-9659,9217,1210,-9352,17225,6331,1506,7124,7939,-4308,-4730,1188,3176,-2475,1424,-48,-469,790,3944,1309,680,1752,2484,1048,341,-282,654,-429,-824,-317,110,-114,-77,514,87,-336,314,152,794,-136,169,-632,188,-172,280,-21,-59,-628,-366,-701,125,-342,195,-484,14,-385,142,-203,-25,-63,-111,-438,-237,-144,-19,-603,-408,-283,-225,-622,-635,-157,-385,-331,-577,-30,-389,-404,-427,-106,-226,-443,-405,-162,-392,-396,-469,17,-462,-322,-529,-39,-465,-308,-396,-19,-390,-307,-364,11,-326,-213,-317,-16,-354,-213,-313,-60,-343,-187,-297,-56,-344,-144,-249,-9,-231,-112,-162,-24,-173,-102,-136,27,-150,-25,-210,56,-219,64,-254,58,-220,23,-244,-12,-203,-8,-244,-39,-226,-35,-259,-51,-231,-34,-263,-76,-247,-60,-275,-44,-227,-27,-295,-46,-242,-22,-251,-17,-192,-18,-252,-5,-165,49,-233,62,-170,73,-217,68,-124,45,-174,23,-102,4,-167,36,-98,28,-187,19,-138,2,-165,-15,-133,-54,-155,-42,-139,-76,-178,-15,-149,-36,-176,14,-154,-30,-174,22,-136,-10,-157,29,-146,8,-148,67,-147,48,-150,87,-144,95,-120,68},
+   /* IRC_Composite_C_R0195_T030_P345.wav */
+   {-5,4,-2,1,1,-3,5,-8,11,-14,18,-22,28,-33,40,-47,52,-40,-408,-1246,-239,-636,412,-1166,-641,-1047,-627,-510,85,1903,1495,1980,-2101,-8149,8335,-7570,2561,15248,-2261,7441,12350,8152,-6960,-897,-1312,-80,35,3139,-1223,-111,782,2694,1229,2248,830,1733,-78,616,211,239,-1106,180,-577,174,-80,366,-414,-235,48,899,-101,-343,251,6,-139,-259,84,-149,-439,-392,-377,-353,-641,-180,-347,-645,-512,18,-444,-412,-353,-372,29,-540,-286,-340,91,-270,81,-387,-257,-287,-105,-191,-258,-436,11,-476,-269,-411,-52,-449,-147,-462,-317,-370,-394,-207,-298,-370,-445,-321,-432,-282,-273,-429,-392,-579,-276,-340,-267,-422,-237,-397,-288,-363,-183,-311,-252,-339,-167,-369,-181,-301,-85,-308,-243,-297,-206,-198,-141,-225,-133,-188,-51,-81,-5,-156,39,-131,-42,-137,-12,-105,-59,-165,-33,-91,-103,-131,-61,-132,-15,-166,-70,-196,-74,-136,-77,-240,-119,-181,-167,-207,-153,-210,-116,-267,-128,-236,-136,-218,-68,-191,-86,-215,-66,-87,-41,-115,-57,-149,39,-73,32,-134,21,-60,58,-96,16,-33,13,-37,-10,-59,0,-62,-11,-108,-26,-126,5,-173,-25,-133,-22,-188,-62,-152,-27,-144,-75,-133,-61,-105,-49,-114,-43,-128,-43,-94,-30,-105,-18,-125,-14,-97,30,-123,23,-65,42,-66,35,-56},
+   /* IRC_Composite_C_R0195_T045_P345.wav */
+   {-8,9,-9,10,-11,12,-14,15,-17,18,-20,23,-26,29,-32,27,-914,-992,248,-1147,746,-600,-1411,-744,-565,-282,187,2409,2683,-221,-1093,-6934,4033,-4276,11276,-297,4328,20553,5592,-1738,-1835,48,-2700,3273,333,-1462,-612,3497,663,1061,1222,2492,754,1168,-36,273,-1174,-272,-1054,891,-844,132,-415,315,-117,257,-335,440,-662,37,-141,164,-657,106,-514,-245,-612,-181,-587,-119,-354,-272,-370,70,-460,-182,-631,-316,-708,-110,-803,-99,-670,-498,-427,-54,-498,-454,-369,-295,-361,-311,-286,-279,-189,-173,-337,-51,-132,-118,-321,-115,-360,-197,-296,-134,-359,-223,-370,-154,-278,-322,-354,-297,-436,-355,-372,-287,-465,-352,-358,-269,-478,-297,-336,-244,-461,-251,-336,-244,-377,-295,-277,-241,-357,-246,-212,-210,-283,-149,-224,-96,-233,-70,-163,-39,-261,-32,-98,-58,-219,-32,-96,-67,-134,7,-101,-50,-139,18,-115,-50,-164,-70,-125,-101,-200,-71,-185,-130,-181,-136,-176,-132,-203,-147,-173,-138,-199,-159,-176,-127,-141,-96,-163,-71,-143,-38,-126,-27,-77,-59,-103,5,-43,-32,-25,-36,-5,-42,40,-34,18,-45,1,-59,7,-47,-28,-49,-55,-63,-33,-55,-79,-59,-82,-109,-94,-40,-81,-89,-121,-82,-83,-49,-111,-62,-79,-97,-59,-52,-74,-97,-50,-98,-31,-116,11,-85,32,-106,43,-88,79,-70},
+   /* IRC_Composite_C_R0195_T060_P345.wav */
+   {2,0,-1,2,-4,7,-10,13,-18,24,-33,46,-69,124,-1066,18,-1103,-58,-112,-921,-201,357,-2572,-292,84,3712,705,1846,-3182,733,-1997,-2694,2428,-595,23701,7754,-2253,5078,3945,-1381,793,-3491,-1163,3714,3613,-2506,296,680,1730,1310,1740,569,508,99,-722,-392,-453,-871,66,-328,-669,-71,-484,-147,-35,146,49,-532,-397,-139,-139,-522,-373,-116,-571,-415,-445,-69,-274,-275,-429,-422,-271,-360,-287,-382,-22,-407,-385,-428,-388,-334,-422,-213,-696,-500,-450,-310,-488,-360,-442,-369,-341,-387,-436,-315,-312,-227,-524,-284,-360,-145,-448,-103,-329,-255,-335,-170,-226,-169,-275,-186,-277,-175,-313,-153,-328,-106,-340,-187,-359,-196,-395,-186,-410,-204,-401,-242,-445,-212,-399,-269,-333,-214,-369,-275,-249,-157,-224,-209,-240,-93,-196,-87,-134,-136,-170,-118,-141,-152,-108,-143,-171,-122,-121,-75,-167,-118,-171,-34,-161,-63,-166,-73,-182,-54,-115,-102,-170,-130,-176,-78,-151,-106,-187,-132,-212,-115,-143,-110,-184,-158,-155,-100,-103,-72,-126,-49,-120,-26,-63,20,-83,6,-50,25,-17,26,-29,39,-31,8,-37,34,-77,20,-77,4,-80,-26,-93,-19,-70,-86,-71,-70,-82,-94,-51,-114,-61,-80,-78,-73,-90,-68,-70,-42,-80,-51,-78,-111,-16,-79,-18,-107,-18,-114,-3,-27,-1,-36,-19,-21,-10,39},
+   /* IRC_Composite_C_R0195_T075_P345.wav */
+   {13,-13,13,-13,13,-13,13,-14,14,-14,14,-14,13,-149,-486,-602,-335,-455,-212,-518,0,-1171,-540,-311,1579,3683,3360,-5772,-426,-1232,-715,5798,-5995,21017,15656,-2523,-3222,5350,1893,1313,-2953,-885,4326,1209,-279,-1161,869,861,1767,953,-85,210,-139,-663,-17,-553,268,-673,-215,-760,-236,-646,-139,-393,-427,-462,-55,-562,-131,-287,-193,-567,-273,-625,-345,-491,-150,-487,-111,-512,-283,-603,-50,-418,-328,-443,-111,-346,-355,-349,-35,-305,-248,-516,-491,-547,-144,-443,-470,-467,-394,-484,-216,-365,-370,-505,-259,-440,-318,-386,-196,-476,-376,-474,-304,-362,-195,-334,-380,-413,-292,-310,-213,-289,-263,-333,-155,-255,-179,-252,-180,-265,-153,-285,-190,-264,-147,-236,-237,-304,-248,-194,-182,-204,-186,-250,-173,-172,-144,-229,-114,-172,-167,-183,-136,-161,-106,-122,-161,-186,-139,-156,-91,-182,-139,-216,-134,-193,-139,-180,-115,-198,-155,-201,-90,-175,-115,-207,-160,-200,-119,-149,-119,-195,-93,-130,-96,-135,-84,-130,-121,-106,-85,-107,-61,-110,-53,-77,2,-45,-15,-51,-20,4,33,-23,-40,-36,1,-53,16,-46,-46,-69,1,-46,-34,-47,-35,-46,-45,-73,-11,-76,-35,-107,-21,-105,-57,-70,-68,-101,-73,-74,-59,-95,-48,-78,-69,-89,-82,-50,-61,-44,-75,-23,-52,-44,10,-43,-23,-47,63,-13,31,19},
+   /* IRC_Composite_C_R0195_T090_P345.wav */
+   {-40,45,-52,60,-69,79,-92,108,-129,156,-197,267,-462,-149,208,-635,-826,268,-390,-920,244,-62,-1260,-320,2937,4521,-564,-4829,2706,-6796,9962,-6783,6356,23745,4526,-4606,-155,5413,3814,-1343,-3207,4343,2939,-733,-2665,998,738,1892,332,-416,-415,112,-618,265,-680,-48,-750,405,-729,216,-673,-99,-730,-121,-855,-310,-693,-146,-680,-198,-695,-260,-522,-283,-523,-143,-597,-333,-173,-342,-516,-211,-519,-364,-344,-136,-671,-213,-411,-202,-398,-103,-578,-290,-294,-138,-306,-230,-402,-283,-321,-289,-337,-289,-473,-414,-279,-358,-413,-275,-431,-453,-313,-307,-448,-301,-341,-461,-354,-322,-417,-358,-269,-383,-284,-242,-302,-305,-257,-285,-308,-182,-268,-314,-270,-212,-228,-190,-249,-276,-158,-168,-143,-161,-146,-207,-91,-103,-158,-110,-126,-88,-73,-13,-110,-35,-46,-106,-57,-116,-140,-215,-98,-197,-197,-174,-246,-265,-202,-185,-227,-181,-218,-214,-125,-146,-175,-178,-153,-217,-80,-87,-137,-151,-98,-122,-94,-88,-136,-141,-168,-115,-121,-70,-143,-102,-56,-76,-36,6,-27,-49,15,48,-26,23,-19,19,-21,34,-59,8,-22,-47,2,-38,-69,-64,-5,-74,-105,-59,-66,-97,-94,-96,-78,-97,-52,-113,-64,-142,-56,-66,-39,-129,-40,-66,-56,-70,5,-96,-30,-55,4,-41,13,-55,20,-26,32,-22,71,-18,17,30},
+   /* IRC_Composite_C_R0195_T105_P345.wav */
+   {-67,70,-74,77,-81,86,-90,96,-101,108,-115,124,-133,69,-792,-387,271,-596,-717,859,-864,-953,-174,2178,-1005,1282,2249,6443,-10856,1162,-3444,7735,12493,-6470,12789,13090,-1110,-7098,1720,6128,6522,-3157,-2418,823,2649,-1545,-82,75,412,176,-227,-1102,-27,27,-108,-705,430,-492,-144,-623,458,-659,-400,-488,-145,-709,-561,-722,-446,-592,-392,-773,-320,-376,-219,-435,-230,-433,-344,-257,-282,-422,-405,-416,-298,-443,-368,-500,-217,-519,-166,-433,-279,-366,-130,-440,-244,-224,-238,-388,-222,-308,-238,-178,-169,-397,-297,-264,-256,-355,-298,-417,-426,-330,-305,-333,-364,-344,-434,-356,-283,-355,-382,-364,-318,-364,-236,-292,-356,-334,-239,-297,-263,-218,-274,-335,-226,-222,-269,-272,-264,-249,-207,-170,-177,-194,-173,-129,-105,-104,-107,-136,-46,-4,64,55,23,-30,-30,-40,-113,-234,-240,-226,-142,-197,-242,-304,-222,-141,-127,-177,-202,-148,-92,-95,-154,-222,-191,-141,-114,-142,-174,-129,-161,-67,-128,-34,-186,-89,-116,-46,-141,-77,-118,-122,-94,-41,-91,-117,-59,-17,-48,-43,-49,-43,-26,26,1,-33,-44,-8,1,24,-53,-52,-67,-16,-53,-12,-77,-55,-119,-25,-52,-50,-119,-50,-99,-48,-93,-38,-125,-90,-97,-23,-111,-77,-96,-23,-117,-28,-57,-25,-82,-14,-37,-3,-37,48,-10,30,-29,94,17,67},
+   /* IRC_Composite_C_R0195_T120_P345.wav */
+   {-3,4,-6,7,-9,12,-14,18,-21,26,-32,40,-51,68,-105,-237,-115,-196,-240,-710,503,-404,-299,-610,1573,-1052,1538,3156,4734,-10274,2428,47,3883,7208,-4454,16464,10228,-4264,-5316,5989,6995,2607,-3203,-668,1640,904,-2642,-469,203,1050,-729,-1586,-33,324,-227,138,-165,-76,50,-374,-903,385,-637,-751,-371,7,-958,-172,-379,-347,-638,-287,-600,-251,-397,-596,-514,-222,-363,-323,-357,-509,-371,-223,-433,-356,-378,-472,-223,-149,-381,-362,-256,-360,-249,-245,-414,-343,-215,-318,-296,-253,-281,-260,-186,-267,-334,-184,-218,-221,-298,-323,-280,-235,-266,-346,-291,-364,-250,-269,-330,-407,-337,-288,-297,-291,-367,-331,-313,-227,-277,-310,-324,-354,-249,-294,-307,-326,-246,-243,-266,-239,-219,-134,-170,-168,-181,-178,-179,-104,-105,-182,-117,-16,21,-1,-16,-123,-97,-70,-125,-190,-191,-224,-174,-191,-187,-240,-111,-186,-142,-146,-92,-134,-84,-176,-166,-110,-109,-146,-139,-231,-119,-73,-70,-193,-83,-143,-72,-63,-63,-163,-69,-90,-96,-97,-84,-157,-37,-91,-88,-116,-29,-99,-31,-73,-68,-76,24,-65,-8,-90,-71,-32,33,-119,-46,-74,-20,-92,37,-144,-45,-100,37,-86,3,-132,2,-42,-29,-104,-4,-152,-16,-85,-19,-175,14,-94,-9,-106,-24,-72,42,-77,-27,-22,2,-33,78,-29,-21,21,111,-5},
+   /* IRC_Composite_C_R0195_T135_P345.wav */
+   {-8,8,-9,9,-9,10,-10,11,-11,11,-12,12,-12,13,-12,12,-14,-199,-345,194,-292,-20,-363,219,-924,441,900,798,-558,3720,1651,-3208,-514,-4991,13814,1350,-1342,10873,8019,1098,-3160,1031,6880,5706,-3917,-4208,1661,708,-1088,-1854,-201,229,-177,-273,-648,-82,475,5,-734,169,-151,-581,-653,471,-491,-236,-587,-109,-475,-248,-581,-317,-480,-514,-555,-293,-524,-197,-571,-212,-393,-202,-577,-116,-375,-344,-418,-218,-355,-263,-348,-305,-348,-210,-371,-214,-394,-179,-361,-184,-347,-189,-348,-251,-234,-293,-267,-213,-252,-242,-269,-147,-370,-183,-325,-201,-333,-191,-269,-230,-243,-283,-228,-273,-293,-252,-342,-267,-359,-215,-428,-224,-365,-239,-330,-269,-348,-257,-251,-286,-250,-246,-272,-177,-242,-155,-277,-116,-236,-94,-218,-96,-109,-104,-71,-63,-76,-143,-84,-111,-151,-139,-218,-149,-229,-159,-185,-165,-192,-165,-156,-158,-137,-140,-159,-114,-167,-123,-164,-97,-148,-122,-112,-140,-90,-115,-78,-111,-104,-86,-114,-49,-142,12,-166,-28,-168,-17,-113,-58,-89,-81,-116,-67,-58,-65,-96,-68,-92,-33,-99,-37,-87,-67,-90,-37,-68,-114,-65,-87,-50,-124,-30,-83,-1,-96,-4,-55,-19,-87,13,-80,-10,-90,-5,-84,-42,-79,4,-67,-63,-38,-27,-36,-23,-7,11,-34,14,-18,51,-74,41,9,19,-7},
+   /* IRC_Composite_C_R0195_T150_P345.wav */
+   {9,-10,11,-12,13,-14,16,-18,20,-22,25,-28,32,-36,41,-48,57,-69,92,-187,-93,-65,148,-471,61,62,-108,-513,1146,660,197,907,4643,-2481,-3069,2384,-436,10331,-1444,3529,10790,4133,-3114,1101,5334,2231,175,-4639,-668,1364,-942,-1832,-179,106,-201,-621,-288,256,-14,7,-70,10,-62,-163,-538,-397,250,-662,-393,-549,-98,-597,-308,-388,-216,-452,-393,-291,-335,-411,-150,-445,-160,-290,-89,-416,-142,-299,-403,-360,-239,-271,-373,-246,-311,-338,-233,-221,-259,-317,-192,-311,-255,-207,-237,-276,-272,-275,-277,-233,-252,-250,-181,-293,-170,-245,-207,-274,-163,-258,-249,-201,-205,-239,-243,-229,-292,-303,-216,-288,-278,-312,-241,-345,-256,-284,-302,-346,-258,-265,-246,-271,-236,-265,-215,-277,-207,-240,-196,-255,-168,-178,-155,-144,-107,-152,-110,-104,-74,-180,-121,-180,-122,-196,-144,-245,-129,-207,-127,-182,-107,-225,-82,-192,-124,-217,-97,-227,-156,-196,-108,-174,-119,-141,-77,-130,-30,-126,-49,-162,-24,-161,-3,-167,-53,-117,-53,-120,-68,-81,-70,-85,-45,-100,-52,-108,-40,-95,-39,-123,-50,-123,-64,-117,-58,-114,-101,-106,-61,-84,-90,-69,-81,-96,-40,-52,-52,-64,-50,-43,-17,-37,-37,-34,-28,-67,-18,-49,-26,-48,-2,-23,-34,-24,7,4,-14,13,9,-1,-21,7,5,3,-11,-9},
+   /* IRC_Composite_C_R0195_T165_P345.wav */
+   {-16,16,-16,16,-16,17,-17,17,-17,18,-18,18,-18,19,-19,19,-19,19,-19,19,-18,15,-25,-180,39,144,-199,84,-114,-19,-188,1254,741,544,587,3703,-1657,-2617,2350,59,8499,1392,3068,7765,4121,-1476,-112,3753,-281,908,-3882,-912,-13,232,-1645,225,-231,30,-284,242,364,101,-416,-192,-215,-224,-211,-588,-413,94,-532,-410,-203,-224,-476,-149,-162,-206,-299,-157,-323,-203,-258,-155,-285,-195,-259,-160,-310,-161,-188,-219,-325,-228,-252,-329,-249,-318,-288,-289,-215,-279,-273,-237,-265,-253,-240,-166,-308,-223,-268,-204,-257,-185,-200,-214,-215,-169,-225,-233,-270,-224,-251,-229,-254,-245,-245,-239,-242,-271,-250,-255,-257,-319,-229,-320,-280,-334,-233,-335,-241,-307,-225,-319,-212,-265,-239,-293,-245,-235,-213,-218,-190,-158,-146,-114,-164,-138,-176,-126,-194,-163,-222,-185,-202,-181,-221,-210,-185,-167,-169,-169,-149,-149,-121,-122,-91,-166,-97,-138,-90,-161,-94,-134,-93,-167,-101,-146,-89,-171,-96,-130,-90,-122,-75,-116,-77,-102,-73,-86,-102,-91,-92,-79,-122,-82,-99,-73,-113,-89,-86,-78,-89,-91,-110,-85,-90,-61,-78,-110,-68,-88,-12,-123,-18,-120,12,-117,12,-119,3,-74,28,-74,-4,-63,7,-46,0,-67,7,-52,21,-52,5,-33,17,5,-5,-18,-6,3,-14,-17,-13,-5},
+   /* IRC_Composite_C_R0195_T180_P345.wav */
+   {14,-14,15,-15,15,-16,16,-17,18,-18,19,-20,20,-21,22,-23,25,-26,28,-31,33,-37,43,-53,84,-38,163,-71,-158,103,194,-92,53,482,91,607,1972,2332,-1572,-66,-260,1180,4049,-804,6508,7047,3077,-817,1693,1744,560,329,-3303,313,-242,-704,-831,249,-87,-229,183,93,110,39,-28,-382,-360,-250,-254,-387,-156,-146,-568,-151,-222,7,-192,-113,-168,-1,-224,-13,-332,-98,-241,-68,-295,-42,-210,-57,-298,-57,-251,-128,-317,-192,-299,-254,-305,-255,-341,-262,-251,-237,-260,-298,-266,-251,-226,-202,-261,-186,-262,-164,-227,-165,-247,-142,-218,-142,-250,-192,-303,-264,-330,-282,-325,-281,-325,-258,-295,-215,-312,-214,-308,-229,-274,-251,-309,-306,-294,-311,-271,-299,-256,-284,-250,-220,-205,-223,-287,-266,-239,-195,-135,-147,-134,-149,-99,-176,-199,-275,-263,-263,-286,-264,-295,-223,-247,-187,-176,-146,-128,-120,-103,-115,-67,-85,-89,-116,-143,-100,-117,-73,-127,-58,-97,-32,-121,-65,-147,-77,-157,-103,-172,-126,-140,-135,-140,-152,-146,-132,-154,-119,-152,-105,-161,-118,-130,-105,-117,-95,-88,-108,-66,-84,-63,-104,-55,-48,-36,-55,-20,-46,-8,-40,5,-55,-25,-61,-13,-59,-44,-51,-47,-33,-43,-29,-37,-41,-40,-27,-26,-28,-32,-23,-2,-7,24,-14,-21,-16,-7,-26,-52,-44},
+   /* IRC_Composite_C_R0195_T195_P345.wav */
+   {6,-6,6,-7,7,-7,7,-7,7,-7,8,-8,8,-8,9,-9,9,-10,10,-11,11,-12,14,-15,17,-21,38,-16,75,-220,-59,106,197,-177,5,-30,-104,721,1540,1146,-1007,1149,558,-528,530,189,5406,5771,1722,1227,2989,2251,-549,199,-1093,649,351,-1192,-797,86,474,-276,161,-98,513,-55,324,-259,35,-250,203,-294,-48,-254,-20,-107,43,-102,74,7,58,5,31,16,-81,-113,-13,-32,-61,-116,-90,-161,-57,-141,-149,-200,-221,-127,-211,-175,-301,-256,-303,-211,-249,-283,-281,-255,-186,-218,-164,-228,-189,-224,-141,-185,-133,-235,-194,-211,-160,-216,-234,-256,-252,-243,-263,-310,-294,-327,-270,-308,-298,-307,-298,-251,-283,-236,-306,-260,-308,-239,-300,-271,-297,-271,-265,-262,-234,-272,-265,-257,-272,-222,-275,-166,-216,-116,-201,-132,-209,-165,-253,-247,-300,-289,-266,-286,-274,-274,-215,-180,-166,-157,-142,-89,-78,-66,-105,-65,-95,-58,-92,-79,-99,-64,-63,-57,-120,-89,-112,-118,-168,-135,-190,-134,-199,-134,-197,-155,-189,-166,-196,-156,-193,-139,-163,-148,-164,-142,-130,-130,-110,-101,-105,-93,-75,-95,-30,-101,-23,-80,-12,-54,-3,-54,-21,-40,-15,-43,-40,-44,-53,-61,-41,-51,-45,-65,-44,-54,-43,-56,-70,-49,-64,-4,-42,4,-66,5,-19,28,-23,-34,-32,-54},
+   /* IRC_Composite_C_R0195_T210_P345.wav */
+   {-2,2,-2,2,-2,2,-2,1,-1,1,-1,1,-1,0,0,0,1,-1,1,-2,3,-3,4,-6,7,-9,13,-20,43,-89,-222,10,103,-107,-34,-36,142,-400,-30,464,582,998,189,-102,357,928,-496,3,2407,3014,4127,2156,97,2572,2367,192,-1077,553,225,383,-471,-511,-191,662,222,22,339,294,362,131,306,286,107,75,262,201,100,126,189,137,172,10,156,70,103,-4,-19,-25,-76,-86,-32,-105,-27,-153,-34,-185,-5,-211,-79,-248,-114,-227,-127,-231,-155,-259,-155,-256,-173,-269,-198,-265,-178,-210,-163,-209,-150,-214,-137,-236,-119,-268,-155,-295,-185,-296,-225,-287,-248,-275,-270,-274,-255,-294,-292,-305,-257,-323,-287,-335,-263,-321,-272,-309,-253,-288,-250,-283,-256,-298,-279,-264,-275,-235,-286,-206,-259,-185,-242,-173,-250,-179,-242,-182,-246,-189,-235,-187,-222,-180,-217,-154,-198,-159,-175,-140,-165,-133,-123,-114,-152,-125,-126,-116,-132,-139,-111,-130,-90,-128,-110,-154,-128,-148,-134,-149,-163,-162,-154,-157,-161,-181,-160,-162,-162,-166,-161,-175,-145,-160,-146,-183,-159,-151,-121,-142,-111,-108,-118,-82,-70,-76,-70,-81,-45,-58,-32,-55,-61,-50,-56,-13,-58,-19,-65,-23,-55,-34,-79,-47,-53,-75,-77,-59,-49,-49,-45,-23,-65,-15,-50,-1,-35,-44,-51,-18},
+   /* IRC_Composite_C_R0195_T225_P345.wav */
+   {4,-4,4,-4,5,-5,5,-5,5,-5,5,-5,5,-5,5,-6,6,-6,6,-6,6,-6,7,-7,7,-7,8,-8,8,-8,9,-22,-68,-102,-39,-33,-2,-125,-43,-65,-81,-88,504,491,271,136,50,242,413,575,246,1154,3270,2343,1173,1360,1386,1335,693,41,-16,626,497,-208,385,-141,534,407,815,196,540,538,608,534,497,553,348,395,314,320,191,167,89,75,-22,80,-26,77,-33,113,-103,86,-85,66,-164,-34,-138,-32,-150,-66,-143,-81,-162,-134,-151,-126,-180,-176,-206,-176,-215,-184,-242,-174,-257,-171,-240,-144,-234,-156,-238,-145,-213,-186,-241,-198,-236,-200,-304,-237,-309,-223,-315,-256,-320,-273,-306,-276,-282,-301,-281,-293,-281,-297,-281,-281,-278,-300,-296,-308,-282,-258,-291,-257,-276,-228,-231,-194,-203,-185,-194,-181,-173,-165,-183,-152,-208,-152,-227,-158,-213,-156,-222,-165,-206,-169,-192,-153,-170,-166,-179,-180,-153,-149,-139,-152,-153,-126,-111,-90,-107,-114,-128,-119,-123,-144,-150,-167,-156,-182,-180,-204,-160,-191,-173,-198,-185,-178,-167,-163,-174,-180,-175,-186,-163,-152,-140,-147,-132,-122,-102,-97,-72,-79,-75,-66,-54,-46,-62,-28,-64,-26,-69,-23,-69,-28,-68,-48,-80,-55,-84,-64,-71,-62,-66,-82,-71,-59,-36,-53,-39,-61,-45,-48,-32,-64},
+   /* IRC_Composite_C_R0195_T240_P345.wav */
+   {1,-1,1,-1,1,-1,1,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,3,-3,3,-3,4,-4,5,-7,11,-87,-91,-69,-34,-17,-83,-130,-72,-103,-104,-134,114,-4,-17,85,373,202,-49,-101,219,365,876,782,785,1628,2056,1505,913,728,668,1183,655,491,465,893,831,430,584,712,1020,649,708,723,730,613,504,566,414,315,187,278,133,226,116,259,129,186,116,185,45,88,-32,30,-121,-4,-138,-59,-242,-77,-196,-88,-164,-95,-154,-158,-144,-123,-166,-155,-131,-163,-193,-126,-158,-197,-221,-177,-191,-199,-207,-207,-232,-259,-213,-257,-223,-245,-210,-232,-214,-199,-250,-237,-259,-286,-308,-300,-276,-316,-306,-323,-315,-283,-322,-259,-334,-262,-308,-246,-279,-237,-252,-210,-235,-204,-211,-204,-216,-219,-187,-208,-184,-188,-172,-182,-147,-178,-171,-209,-148,-207,-140,-214,-153,-202,-132,-178,-141,-158,-129,-140,-132,-148,-104,-170,-94,-160,-103,-179,-127,-186,-148,-195,-164,-196,-178,-176,-178,-191,-206,-200,-201,-208,-201,-208,-200,-195,-166,-167,-160,-164,-138,-176,-171,-170,-145,-157,-125,-127,-91,-104,-77,-96,-65,-75,-71,-96,-65,-84,-60,-84,-57,-72,-49,-85,-51,-85,-52,-82,-51,-93,-48,-69,-42,-81,-54,-69,-69,-69,-62,-61,-61,-54,-82,-67},
+   /* IRC_Composite_C_R0195_T255_P345.wav */
+   {1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,3,-3,5,-29,-81,-145,-126,-102,-79,-99,-83,-32,26,-25,-104,-85,-9,-133,-67,-57,165,465,613,590,447,814,1074,1050,1231,1480,1384,1691,1545,721,693,1027,967,911,897,961,823,595,850,817,872,522,568,476,335,222,147,215,189,255,142,248,142,175,30,4,8,-59,-17,-77,-20,-108,-60,-97,-148,-132,-187,-115,-138,-127,-111,-127,-113,-156,-101,-162,-102,-180,-108,-179,-120,-179,-195,-199,-229,-185,-270,-215,-278,-237,-258,-241,-261,-210,-244,-198,-260,-213,-287,-247,-340,-255,-344,-240,-343,-221,-330,-201,-299,-199,-288,-250,-262,-234,-258,-219,-245,-213,-243,-209,-226,-210,-206,-229,-185,-199,-171,-204,-168,-186,-151,-175,-154,-197,-147,-168,-130,-180,-149,-171,-122,-158,-129,-152,-132,-156,-139,-160,-164,-170,-174,-154,-162,-126,-151,-134,-162,-148,-177,-191,-199,-212,-200,-209,-220,-215,-220,-217,-209,-214,-187,-188,-163,-169,-145,-176,-134,-160,-141,-157,-132,-139,-118,-136,-115,-131,-104,-132,-107,-134,-96,-102,-76,-89,-80,-83,-80,-72,-65,-58,-64,-74,-61,-66,-46,-79,-66,-94,-77,-85,-74,-86,-77,-80,-67,-80,-78,-81,-61,-58},
+   /* IRC_Composite_C_R0195_T270_P345.wav */
+   {1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,2,-2,2,-3,3,-3,3,-3,4,-4,5,-5,7,-13,-9,25,107,159,235,223,228,197,135,148,180,104,249,332,345,533,679,812,953,1029,929,1077,1230,1424,1718,1696,1725,1631,997,986,1012,146,-166,111,353,347,-6,-43,73,193,90,62,-93,-92,-338,-310,-405,-336,-407,-441,-401,-511,-347,-531,-317,-432,-349,-454,-400,-327,-440,-322,-442,-318,-433,-382,-424,-384,-375,-393,-396,-458,-428,-480,-383,-434,-382,-464,-371,-432,-337,-394,-307,-399,-335,-417,-322,-442,-324,-436,-303,-449,-304,-409,-298,-363,-284,-317,-307,-308,-323,-288,-319,-293,-322,-285,-281,-248,-239,-242,-269,-244,-286,-239,-275,-228,-281,-236,-270,-215,-230,-183,-189,-147,-168,-142,-140,-138,-148,-145,-122,-112,-122,-100,-101,-92,-130,-111,-155,-125,-154,-111,-187,-117,-190,-95,-171,-106,-163,-109,-148,-117,-153,-132,-173,-135,-163,-114,-185,-122,-191,-116,-168,-92,-150,-108,-141,-94,-122,-95,-150,-97,-145,-88,-132,-65,-125,-53,-97,-33,-83,-12,-48,-5,-27,7,-11,35,-15,26,-18,26,-29,3,-32,-9,-39,15,11,7,13,12,6,12,11,20,5,2,4,3,-12,-24,-23,-5,-3,-22,4,5,1,4},
+   /* IRC_Composite_C_R0195_T285_P345.wav */
+   {0,0,0,0,0,0,0,0,0,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-2,2,-2,2,-2,3,-3,3,-4,5,-6,8,-11,22,-196,-173,-113,-87,-175,-179,-137,-214,-236,-209,-197,-203,-89,115,94,63,76,-21,-123,-58,550,837,1139,1631,2133,1768,1607,1992,1501,200,482,1172,950,840,796,772,845,1106,940,1271,764,758,561,601,592,456,196,274,22,269,50,133,94,119,18,4,153,30,97,24,85,20,33,14,-38,-26,-152,-46,-172,-60,-155,-115,-148,-198,-89,-195,-112,-261,-100,-214,-212,-183,-197,-150,-219,-140,-179,-164,-230,-178,-187,-230,-226,-257,-158,-281,-201,-228,-189,-261,-210,-220,-259,-267,-256,-233,-276,-267,-249,-227,-294,-254,-243,-276,-267,-297,-257,-292,-260,-305,-245,-279,-264,-243,-230,-212,-214,-151,-148,-130,-127,-88,-74,-103,-75,-88,-67,-122,-69,-96,-106,-123,-96,-146,-128,-172,-140,-152,-152,-190,-154,-179,-153,-191,-155,-209,-127,-197,-114,-197,-134,-193,-131,-195,-130,-172,-126,-176,-158,-191,-169,-206,-154,-203,-183,-230,-172,-205,-185,-204,-179,-196,-157,-166,-116,-150,-115,-126,-82,-123,-71,-85,-83,-83,-68,-72,-63,-78,-79,-49,-75,-79,-92,-70,-98,-66,-95,-75,-105,-104,-85,-82,-92,-105,-73,-107,-87,-82,-75,-79,-91},
+   /* IRC_Composite_C_R0195_T300_P345.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,3,-3,3,-3,4,-4,5,-6,8,-10,16,-33,4,109,115,258,119,96,128,286,309,258,544,795,778,461,339,836,601,406,1682,1975,1154,2276,2834,1223,701,931,69,-668,-254,489,635,-539,173,339,531,396,94,191,-102,150,-127,-75,-251,-176,-409,-489,-562,-686,-623,-666,-571,-644,-520,-569,-434,-437,-442,-489,-355,-376,-438,-545,-399,-497,-467,-538,-479,-549,-412,-479,-420,-458,-466,-435,-392,-372,-360,-342,-413,-353,-363,-348,-419,-344,-451,-370,-415,-367,-468,-389,-500,-406,-491,-367,-474,-360,-481,-350,-435,-296,-421,-285,-402,-267,-363,-222,-350,-221,-334,-215,-270,-177,-235,-146,-214,-118,-175,-89,-140,-81,-133,-96,-92,-54,-91,-56,-86,-58,-64,-32,-51,-40,-76,-57,-55,-36,-113,-96,-83,-105,-91,-129,-152,-109,-131,-124,-142,-150,-156,-119,-136,-143,-152,-125,-140,-130,-126,-129,-99,-125,-119,-110,-121,-109,-118,-102,-117,-98,-117,-101,-93,-114,-105,-94,-69,-112,-75,-86,-60,-49,-25,-59,6,-33,-1,-8,8,-2,17,23,0,29,13,17,24,18,21,18,23,-13,11,-8,15,5,10,-10,8,-24,7,1,-11,1,27,-4,48,5,53,24,86,43,68,50,99},
+   /* IRC_Composite_C_R0195_T315_P345.wav */
+   {1,-1,1,-1,1,-1,1,-1,1,-1,1,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-1,0,6,-137,-110,-314,-483,-264,-184,-467,-482,89,-336,204,277,141,-93,-770,-71,819,-837,1577,3127,1312,2383,3512,2495,-80,350,890,699,71,561,954,294,1113,1117,1259,495,984,849,877,553,746,524,537,393,441,428,542,429,296,210,204,231,133,-55,33,-7,-111,-11,-91,34,-190,-56,-131,39,-14,66,-79,58,-96,105,-60,169,-64,84,-143,118,-100,-84,-157,-91,-304,-233,-224,-201,-251,-259,-309,-306,-307,-286,-299,-265,-407,-357,-301,-345,-348,-313,-294,-342,-334,-338,-298,-266,-288,-261,-298,-254,-268,-227,-236,-260,-244,-223,-227,-260,-233,-195,-202,-171,-228,-156,-170,-114,-170,-107,-146,-105,-89,-96,-83,-40,-73,-59,-66,-65,-71,-53,-107,-74,-75,-92,-122,-102,-105,-87,-130,-86,-139,-115,-142,-131,-148,-147,-160,-144,-151,-139,-173,-139,-186,-126,-170,-147,-194,-137,-189,-141,-184,-162,-186,-151,-208,-180,-195,-172,-186,-165,-204,-165,-157,-155,-139,-132,-141,-108,-126,-89,-110,-98,-101,-113,-94,-111,-133,-99,-126,-121,-126,-97,-115,-103,-122,-84,-114,-58,-120,-106,-109,-97,-103,-75,-128,-91,-104,-73,-91,-53,-59,-46,-63,-33,-22,-14},
+   /* IRC_Composite_C_R0195_T330_P345.wav */
+   {-9,9,-9,9,-9,9,-9,9,-9,9,-9,9,-9,9,-9,9,-9,9,-9,10,-10,10,-10,10,-10,10,-10,10,-7,-378,-458,118,-356,-539,-320,-251,-977,-150,272,171,262,290,-552,-354,-2031,2393,-54,-667,4888,3245,1575,3792,4173,-210,-907,432,1658,-334,338,623,940,723,1598,1314,1216,764,1038,976,401,116,250,473,239,334,528,294,340,65,340,194,348,152,304,92,279,123,93,-24,89,-192,-125,-56,-130,68,194,-86,77,174,220,128,98,-234,-216,-168,-160,-251,-283,-410,-399,-240,-227,-363,-327,-361,-302,-370,-271,-312,-309,-428,-402,-318,-223,-266,-304,-253,-264,-272,-246,-203,-278,-299,-319,-254,-236,-260,-281,-188,-203,-212,-220,-185,-220,-178,-230,-207,-215,-227,-248,-182,-197,-163,-179,-101,-143,-112,-138,-84,-115,-82,-111,-83,-102,-83,-51,-69,-49,-89,-18,-67,-73,-108,-72,-63,-79,-111,-167,-94,-130,-99,-166,-109,-181,-121,-156,-124,-154,-131,-162,-169,-165,-170,-136,-151,-159,-185,-163,-122,-139,-141,-175,-141,-148,-148,-172,-157,-130,-154,-150,-148,-142,-153,-141,-116,-159,-97,-155,-91,-162,-92,-141,-53,-134,-97,-132,-68,-111,-97,-127,-98,-128,-105,-137,-81,-141,-76,-120,-73,-136,-97,-105,-48,-102,-73,-108,-56,-97,-41,-75,-56,-70,-20,-33,-9,-51,-32},
+   /* IRC_Composite_C_R0195_T345_P345.wav */
+   {1,-2,2,-3,4,-4,5,-6,7,-8,9,-10,11,-13,14,-16,17,-19,21,-22,24,-25,25,-22,13,15,-152,-764,140,-602,18,-900,-519,-1092,-545,489,261,706,-1198,1895,-1912,-2689,2900,30,-3177,8188,3071,2405,4243,6073,615,-3233,507,1858,-510,-537,1688,110,230,1724,2185,1547,1263,1871,1145,573,342,-23,261,-614,-9,-121,805,-137,38,285,272,-29,312,280,184,234,346,-32,603,80,194,153,323,-216,458,198,10,-275,28,-275,-40,-248,-266,-416,-305,-349,-213,-336,-338,-460,-267,-400,-247,-403,-303,-379,-264,-460,-298,-347,-227,-420,-309,-405,-246,-238,-250,-319,-267,-250,-266,-221,-240,-318,-184,-236,-175,-305,-161,-225,-161,-213,-190,-219,-158,-206,-155,-198,-165,-211,-169,-178,-141,-180,-148,-191,-154,-191,-120,-222,-141,-171,-68,-131,-104,-142,-95,-72,-93,-71,-91,-94,-90,-92,-83,-97,-91,-117,-108,-109,-102,-101,-101,-116,-95,-82,-119,-115,-158,-119,-137,-108,-151,-151,-163,-123,-125,-148,-167,-118,-168,-149,-196,-146,-169,-130,-179,-147,-164,-153,-114,-109,-119,-128,-121,-82,-111,-89,-98,-81,-102,-105,-71,-85,-85,-116,-85,-98,-76,-124,-102,-118,-107,-84,-126,-141,-114,-78,-80,-110,-127,-121,-93,-112,-96,-109,-98,-90,-55,-53,-66,-54,-41,-26,-18,-19,-5,-33,-57,-33}};
+
+const int16_t irc_composite_c_r0195_p000[][256] =
+  {/* IRC_Composite_C_R0195_T000_P000.wav */
+   {1,-1,1,-1,0,0,0,0,0,1,-1,2,-2,2,-3,4,-5,6,-7,9,-11,14,-20,33,-756,-534,-348,102,-1032,-582,-1506,65,-959,133,-210,1928,576,-291,-5368,5774,-6962,-2265,14308,-3483,7090,10599,2076,-2731,3149,-192,-3412,-1652,3866,548,-1226,-1088,2012,2077,1960,-209,1087,1404,1871,1150,1665,536,626,1200,1268,592,-114,-160,-95,133,-37,-246,158,-144,-190,-548,421,96,105,-192,181,129,158,143,73,-164,-261,-64,-147,-128,-55,-166,-60,98,349,-253,126,-172,110,-391,-118,265,-317,256,-97,416,-62,274,-244,-355,-104,-427,-357,-741,-368,-489,-365,-418,-498,-150,-384,-361,-530,-257,-297,-381,-491,-511,-393,-429,-294,-395,-373,-408,-292,-342,-260,-297,-286,-427,-324,-291,-291,-276,-319,-244,-386,-221,-241,-141,-260,-273,-231,-153,-86,-203,-202,-152,-95,-140,-176,-127,-110,-156,-237,-148,-139,-94,-248,-190,-202,-85,-155,-205,-223,-169,-158,-183,-216,-167,-164,-146,-219,-103,-162,-71,-220,-10,-169,7,-134,32,-76,-42,-100,-66,-50,-44,-71,-88,-122,-40,-75,21,-78,-18,-125,-33,-69,53,-77,-22,-138,-41,-62,-55,-92,-83,-99,-115,-150,-17,-53,-12,-184,-50,-96,28,-81,-1,-78,-41,-118,-14,-66,38,-82,-29,-104,29,5,34,-62,14,7,47,-61,67,-43,51,-145,61},
+   /* IRC_Composite_C_R0195_T015_P000.wav */
+   {123,-125,127,-129,131,-133,135,-137,138,-140,141,-141,141,-140,137,-131,122,-106,76,-10,-238,-830,-319,-1069,17,-859,165,-1424,-309,-2002,526,695,3378,-155,-423,-6705,9134,-12025,1719,14823,-7846,16119,11510,-1243,-4202,5454,-1293,-2996,-1156,2894,-853,-440,-59,2319,814,827,428,2620,1829,1181,817,1748,469,1133,599,1385,-490,316,-1523,369,-621,391,-835,160,-740,77,-149,751,-195,-103,-622,191,-100,342,-303,-90,-465,82,62,259,-409,-385,44,87,-50,21,54,-467,-207,-79,-324,-108,-90,-152,-620,71,-110,47,-749,-161,-523,-153,-446,-157,-336,40,-137,-86,-507,-177,-322,-14,-430,-277,-557,-360,-326,-296,-274,-463,-359,-502,-263,-388,-316,-369,-353,-459,-464,-335,-293,-324,-333,-369,-324,-380,-288,-268,-287,-294,-317,-270,-395,-181,-204,-149,-383,-214,-212,-113,-188,-174,-211,-188,-99,-166,-173,-203,-111,-134,-148,-226,-166,-126,-86,-216,-199,-234,-119,-193,-118,-250,-172,-245,-101,-156,-114,-172,-153,-162,-98,-70,-71,-106,-70,-81,-14,-14,37,-38,-36,-107,-3,-9,52,-91,-63,-116,20,-50,18,-151,-37,-125,57,-97,31,-149,5,-133,4,-141,9,-132,-7,-168,9,-124,33,-144,22,-166,15,-117,46,-156,11,-127,36,-124,23,-58,69,-65,71,-34,110,-36,130,-45,96,-68,96,-59,38,-71},
+   /* IRC_Composite_C_R0195_T030_P000.wav */
+   {-63,66,-69,73,-76,80,-84,89,-93,98,-102,107,-110,111,-108,95,-54,-154,-785,-302,-1153,268,-1139,240,-1422,320,-2283,272,-629,4046,237,3112,-7137,2285,231,-13166,20036,-5068,11914,16533,2581,-5753,2916,1267,-3077,-1961,1642,-123,210,665,1080,1069,485,1122,1846,1948,901,1486,921,1132,470,865,-186,-57,-358,-153,-489,-118,-724,-365,-517,-156,-335,155,-255,96,-315,376,-169,22,-435,5,-544,-35,-484,-12,-457,280,-453,-177,-399,435,-423,-188,-484,161,-217,-38,-732,-416,-157,73,-480,-440,-302,-140,-260,-335,-490,-420,-470,-291,-617,-112,-315,20,-559,-210,-412,154,-162,-102,-453,-225,-205,-23,-247,-203,-409,-147,-366,-147,-367,-150,-477,-326,-502,-222,-390,-294,-498,-249,-423,-313,-510,-239,-382,-176,-516,-351,-496,-92,-337,-150,-462,-212,-366,-74,-288,-94,-322,-110,-334,-53,-240,11,-306,-108,-332,-45,-271,-52,-295,-76,-308,-99,-319,-57,-226,-50,-303,-130,-254,-54,-183,-89,-261,-98,-193,-6,-184,-12,-164,45,-175,27,-120,129,-69,82,-89,120,-26,113,-48,59,-77,18,-85,-3,-107,0,-78,-29,-159,0,-62,67,-139,-9,-138,19,-136,-3,-124,-27,-149,-30,-136,-11,-97,-7,-137,-19,-96,20,-84,15,-64,28,-92,9,15,121,-12,24,-32,74,36,98,-9,3,-30,36,-42,4},
+   /* IRC_Composite_C_R0195_T045_P000.wav */
+   {72,-74,77,-79,82,-86,91,-96,102,-111,121,-134,152,-179,228,-635,490,-1252,-114,-1202,742,-1779,495,-1563,532,-2895,2675,1167,4718,-4709,4646,-5559,-6673,7608,-8298,21201,11442,4596,920,5308,-1818,-1353,-1082,-283,-725,2349,-984,-140,614,1696,1859,2003,557,2080,1067,416,1112,525,38,-87,-851,-336,38,-375,-495,-291,-403,-90,-299,-393,-130,-647,-37,-270,-227,-182,-2,-485,-21,-380,-221,-179,-364,-244,-343,-247,-161,-213,-443,-201,-386,-349,-198,-453,-401,-479,-187,-349,-331,-486,-40,-343,-132,-248,-346,-150,-185,-498,-200,-280,-512,-271,-152,-472,-405,-421,-235,-444,-272,-369,-299,-285,-209,-483,-192,-140,-287,-241,-120,-328,36,-232,-275,-345,-26,-488,-78,-319,-205,-488,22,-390,-294,-391,-131,-450,-198,-416,-347,-391,-95,-530,-266,-359,-162,-403,-129,-438,-172,-313,-124,-327,-146,-338,-102,-245,-121,-328,-124,-244,-79,-277,-128,-320,-47,-241,-144,-306,-70,-274,-85,-274,-147,-219,-71,-212,-78,-214,-60,-127,0,-182,-21,-106,67,-97,35,-113,84,16,65,-67,99,-24,79,7,29,-85,99,-35,17,-91,57,-100,58,-112,-18,-130,73,-160,-25,-148,6,-136,45,-217,2,-142,79,-180,64,-198,81,-156,114,-200,98,-162,123,-120,152,-170,176,-85,130,-107,189,-99,158,-113,138,-127,161,-140,111,-221},
+   /* IRC_Composite_C_R0195_T060_P000.wav */
+   {123,-132,142,-153,166,-180,196,-215,238,-264,296,-338,397,-545,-26,-1070,548,-1189,312,-1132,-540,-593,703,-1932,23,1834,2821,3650,-4896,2244,-8990,3547,-3667,11688,19991,4002,-67,5265,-1066,-618,257,-237,-93,2490,-1554,-1096,1601,944,1501,1048,1374,1754,1792,566,335,-350,932,-271,-1525,-771,-646,-466,146,-610,113,-1035,510,-222,-23,-608,-443,-725,203,-887,-434,-589,-19,-438,-40,-610,14,-467,-137,-372,-214,-629,-55,-580,-263,-517,-350,-563,-317,-499,-281,-660,-233,-369,6,-533,-180,-457,-90,-331,-140,-395,-150,-246,-124,-225,-134,-388,-151,-322,-330,-363,-280,-394,-241,-503,-138,-532,-169,-402,-263,-433,-255,-363,-244,-372,-271,-364,-268,-337,-126,-472,-58,-441,6,-373,-75,-345,-91,-290,-171,-369,-128,-311,-189,-312,-183,-241,-230,-309,-78,-303,-168,-337,-178,-236,-164,-270,-238,-313,-113,-261,-195,-270,-208,-224,-136,-312,-153,-332,-79,-287,-193,-257,-149,-227,-124,-297,-82,-209,-107,-181,-87,-173,-18,-146,9,-126,28,-74,-22,-25,22,-50,40,-41,25,-17,-14,5,9,-67,23,-34,57,-46,5,3,37,-39,6,-36,11,-54,-39,-69,-61,-79,-35,-99,-38,-98,-29,-88,-12,-86,-19,-71,-19,-37,-12,-58,7,-41,34,-37,47,-22,42,-26,97,2,68,-13,74,7,35,-6,34,-1,0,-38},
+   /* IRC_Composite_C_R0195_T075_P000.wav */
+   {30,-35,41,-47,55,-63,72,-84,97,-113,134,-162,202,-472,-304,-372,274,-1176,-371,-433,-535,-54,364,-1802,1909,3073,5773,-6217,-2323,-1631,-4047,10163,-533,23315,9574,-7064,764,4451,279,-686,938,2910,-1594,-781,31,2302,-241,1013,1336,1127,1162,699,28,761,380,-426,-314,-917,-750,-202,-732,-632,-472,-517,-73,-213,-19,-699,-215,-440,-368,-595,-354,-716,-262,-604,-448,-306,-492,-226,-481,-404,-393,-315,-472,-290,-442,-401,-227,-459,-350,-606,-194,-594,-308,-438,-436,-423,-217,-461,-220,-298,-300,-185,-218,-342,-120,-282,-142,-260,-82,-276,-46,-282,-219,-258,-93,-447,-193,-418,-296,-377,-249,-479,-177,-358,-304,-416,-185,-366,-179,-486,-287,-406,-134,-512,-167,-454,-201,-352,-195,-386,-169,-289,-223,-334,-186,-280,-123,-302,-167,-268,-36,-301,-139,-264,-133,-195,-111,-276,-131,-175,-123,-195,-148,-226,-77,-211,-165,-231,-123,-257,-132,-285,-135,-289,-147,-278,-141,-274,-166,-238,-154,-189,-134,-212,-101,-127,-6,-142,-21,-106,41,-81,41,-105,54,-54,6,-41,-3,-62,27,-57,-25,-42,29,-39,10,-90,56,-38,43,-106,38,-63,27,-58,-21,-110,-19,-82,-36,-124,-57,-83,23,-113,9,-73,55,-109,34,-55,45,-13,62,-54,27,-33,52,12,35,-34,72,-19,33,-32,74,-59,39,-25,53,1,10,-3},
+   /* IRC_Composite_C_R0195_T090_P000.wav */
+   {-3,3,-4,5,-6,7,-9,11,-13,17,-23,36,-967,38,-432,430,-1179,327,-1275,279,-1297,1913,-1800,360,826,8131,-4106,182,-6660,-1308,8763,-6147,21004,13661,-1493,-3341,815,6070,1330,-1696,3689,487,-871,-2812,2923,1220,319,-441,2019,161,1147,-214,819,-1063,179,-1133,125,-1120,701,-1109,-441,-308,-3,-1227,135,-661,-529,-162,-375,-951,-262,-371,-533,-415,-491,-644,-459,-451,-508,-551,-360,-465,-434,-399,-416,-458,-204,-423,-358,-363,-290,-434,-346,-447,-174,-475,-424,-400,-239,-439,-175,-468,-418,-188,-90,-409,-214,-233,-225,-147,-140,-439,-299,-104,-157,-229,-145,-201,-232,-109,-193,-310,-284,-271,-306,-318,-342,-330,-320,-362,-208,-295,-346,-295,-212,-394,-231,-297,-357,-348,-177,-356,-301,-275,-291,-319,-184,-255,-307,-178,-245,-228,-206,-194,-280,-137,-218,-191,-183,-203,-186,-105,-193,-151,-164,-132,-164,-77,-203,-113,-167,-192,-164,-75,-239,-194,-160,-166,-186,-77,-232,-182,-148,-63,-204,-92,-164,-69,-54,4,-140,-18,-43,36,-61,41,-116,-2,-92,42,-126,-61,-150,-8,-117,-15,-62,-69,-114,61,-14,10,-74,8,-25,119,-37,0,-69,10,-56,37,-94,-111,-86,30,-107,-71,-119,4,-92,38,-125,29,-50,58,-57,64,-84,92,-4,67,-73,54,-8,88,-67,71,21,64,-36,123,-7,23,-36,115},
+   /* IRC_Composite_C_R0195_T105_P000.wav */
+   {67,-63,58,-51,43,-33,19,-1,-24,61,-121,244,-927,183,-686,327,-496,-330,-486,-340,-314,396,182,-1212,1259,6063,-556,-2378,-2582,-3838,9614,-8512,18557,14219,-782,-2750,529,5714,4180,-1314,2811,874,-1413,-2611,2334,2590,-537,-811,436,1284,515,-112,-879,-106,-221,-558,-955,-184,-84,-312,-553,-219,-379,-172,-456,-790,-576,-350,-708,-606,-553,-357,-404,-422,-537,-414,-291,-515,-518,-625,-263,-417,-492,-719,-360,-360,-368,-573,-481,-304,-142,-306,-536,-356,-164,-154,-398,-429,-388,-147,-288,-315,-532,-144,-267,-143,-527,-186,-329,6,-447,-226,-375,20,-329,-159,-444,-77,-329,-143,-450,-93,-327,2,-390,-87,-374,55,-418,-96,-445,-37,-436,-129,-505,-118,-444,-152,-487,-172,-395,-135,-432,-170,-368,-96,-414,-145,-394,-32,-326,-123,-434,-83,-276,-94,-355,-149,-242,-56,-256,-166,-269,-75,-228,-119,-313,-87,-207,-58,-304,-97,-234,-23,-285,-126,-259,-11,-242,-91,-247,-1,-204,-10,-242,-5,-194,74,-169,102,-92,170,-62,135,-93,108,-134,5,-190,-8,-203,-47,-223,16,-187,1,-174,58,-131,78,-113,75,-130,98,-75,44,-121,101,-58,53,-146,47,-70,70,-128,-19,-108,21,-89,-23,-119,-27,-39,31,-124,-51,-37,65,-45,0,4,66,6,17,11,14,30,64,44,27,-16,79,-9,-11,-60,55},
+   /* IRC_Composite_C_R0195_T120_P000.wav */
+   {-76,75,-75,74,-72,69,-65,59,-49,35,-14,-21,85,-263,347,27,-533,-67,-385,-42,-335,645,-2065,1305,1033,1922,-2255,6706,-3555,1838,-9048,1510,15767,522,6485,7416,2582,-1635,-1123,7704,6890,-3746,-2349,-815,4058,-1305,633,-1081,2060,-1086,490,-1435,1038,-465,-103,-1558,594,-576,-246,-797,-411,-729,216,-627,-444,-642,122,-670,-147,-796,-255,-906,76,-717,-449,-584,-119,-685,-337,-606,-352,-326,-478,-502,-420,-468,-445,-428,-466,-351,-293,-513,-155,-443,-243,-371,-82,-573,-58,-388,-209,-303,-195,-394,-165,-220,-376,-192,-262,-255,-289,-234,-278,-284,-271,-328,-227,-325,-185,-358,-162,-289,-168,-301,-148,-278,-221,-237,-227,-212,-236,-164,-236,-223,-147,-209,-256,-222,-209,-360,-125,-328,-206,-254,-219,-307,-162,-296,-276,-174,-324,-187,-268,-155,-286,-159,-317,-134,-252,-195,-181,-236,-204,-162,-148,-226,-141,-227,-132,-141,-218,-189,-165,-114,-288,-111,-256,-76,-252,-50,-308,-80,-158,-63,-217,-126,-102,12,-10,-77,13,74,33,49,-22,50,-56,57,-177,-13,-201,19,-144,-101,-181,59,-32,-49,-45,-72,-16,32,-40,-48,-33,-42,-45,64,-145,8,-69,33,-144,44,-92,56,-118,-2,-60,8,-77,-3,-91,-37,-73,-11,-53,0,-49,56,-45,17,-73,100,6,64,-7,125,-47,94,22,33,-55,12,-30,7},
+   /* IRC_Composite_C_R0195_T135_P000.wav */
+   {-47,46,-46,44,-43,40,-37,33,-28,21,-12,-1,20,-48,94,-206,95,-290,-258,153,-592,203,-293,297,-1153,1348,644,1791,-1433,6094,-3856,-260,-5383,6042,11744,-1699,5991,8208,1865,-2201,2510,6226,4590,-5126,-1361,782,2433,-2030,854,-833,853,-1117,568,-1205,680,-1070,-374,-797,196,-738,-381,-460,54,-509,80,-809,-117,-538,-17,-916,203,-876,-170,-815,-257,-782,-178,-520,-324,-433,-344,-353,-383,-382,-403,-522,-240,-512,-312,-523,-189,-533,-114,-477,-247,-366,-173,-363,-247,-204,-351,-273,-220,-239,-329,-247,-257,-330,-207,-331,-212,-342,-200,-298,-246,-261,-277,-228,-264,-213,-328,-212,-273,-219,-256,-244,-234,-265,-159,-264,-191,-244,-157,-229,-190,-221,-203,-203,-265,-198,-238,-189,-235,-164,-189,-220,-183,-216,-189,-257,-194,-227,-199,-234,-209,-214,-214,-195,-212,-179,-255,-200,-207,-151,-257,-182,-194,-172,-200,-169,-219,-215,-208,-180,-207,-192,-215,-173,-203,-137,-219,-101,-232,-121,-175,-4,-148,-45,-92,5,-35,0,-5,-45,0,8,10,-53,-107,-5,-81,6,-157,3,-130,19,-128,39,-138,5,-65,57,-90,7,-52,45,-79,13,-78,29,-109,14,-79,25,-126,7,-61,-11,-75,1,-51,-46,-64,-35,-74,-40,-48,36,-46,3,-64,53,-59,33,-26,105,4,63,13,64,-28,24,-1,3,-80,-21},
+   /* IRC_Composite_C_R0195_T150_P000.wav */
+   {-5,6,-7,7,-8,9,-10,11,-12,13,-15,16,-17,19,-20,20,-17,1,-234,227,-421,117,-50,-175,-40,33,-222,178,1388,586,910,1027,4228,-7050,1223,2017,6153,4669,1121,8628,6102,-3597,1339,6488,2001,-1618,-2670,715,665,-9,-1202,163,-501,22,-977,268,-483,330,-922,182,-469,62,-791,-223,-334,-215,-485,-347,-491,-176,-429,-178,-696,-63,-705,-191,-727,-100,-715,-18,-605,-161,-531,-137,-475,-197,-485,-201,-433,-135,-408,-232,-404,-136,-441,-131,-469,-151,-475,-144,-469,-155,-431,-165,-416,-207,-381,-199,-342,-208,-344,-172,-301,-189,-339,-193,-297,-159,-303,-169,-283,-144,-295,-190,-323,-198,-312,-172,-324,-214,-279,-169,-264,-159,-230,-157,-232,-160,-206,-167,-240,-200,-232,-191,-234,-175,-238,-189,-225,-142,-206,-214,-206,-200,-152,-232,-176,-232,-184,-217,-152,-205,-194,-188,-192,-160,-199,-163,-211,-186,-234,-194,-218,-212,-270,-194,-202,-181,-219,-182,-202,-169,-194,-142,-192,-142,-138,-58,-121,-63,-104,-7,-62,-12,-40,0,-55,-16,-33,-41,-87,-38,-71,-48,-89,-7,-81,-22,-59,5,-41,-28,-47,-20,-14,-23,-20,-10,-31,-22,-25,-11,-55,-34,-32,-33,-44,-53,1,-30,-56,-58,-29,-61,-47,-27,-44,-54,-52,-32,-37,-63,-16,18,1,-3,18,72,0,21,9,42,-23,-7,-40,-42},
+   /* IRC_Composite_C_R0195_T165_P000.wav */
+   {-31,31,-31,31,-30,30,-30,29,-28,27,-25,23,-20,16,-12,5,5,-18,40,-78,187,-74,-49,55,-110,112,-199,218,-425,530,199,1304,62,2352,168,2823,-5315,837,4953,2252,4624,4623,6361,3213,-1540,938,5698,-601,-1840,-1321,415,-514,-55,-892,3,-75,186,-753,471,-260,192,-752,21,-345,-220,-374,-258,-349,-282,-436,-427,-398,-253,-570,-291,-449,-37,-524,-88,-564,-148,-493,-85,-435,-73,-317,-105,-359,-181,-335,-176,-321,-195,-365,-206,-391,-180,-383,-187,-437,-200,-456,-199,-434,-221,-404,-185,-335,-239,-304,-221,-288,-247,-317,-222,-256,-192,-283,-227,-290,-193,-268,-185,-298,-185,-261,-149,-305,-194,-283,-162,-272,-207,-271,-209,-214,-185,-206,-201,-195,-156,-200,-178,-267,-179,-273,-180,-271,-149,-245,-175,-193,-143,-182,-204,-213,-175,-213,-191,-252,-204,-249,-193,-210,-188,-202,-194,-140,-186,-146,-188,-134,-188,-186,-187,-199,-210,-191,-202,-211,-250,-183,-225,-176,-254,-154,-188,-96,-150,-85,-102,-69,-80,-63,-71,-81,-74,-55,-81,-81,-69,-57,-79,-51,-37,-34,-57,-26,-18,6,-12,5,-5,22,-15,19,-13,6,-49,9,-38,-32,-76,-25,-65,-43,-75,-1,-82,-14,-96,-21,-95,-25,-47,-4,-67,-76,-41,-67,-39,-53,-5,-25,-37,30,14,61,-21,72,-39,30,-59,25,-48,-21},
+   /* IRC_Composite_C_R0195_T180_P000.wav */
+   {9,-9,9,-10,10,-11,11,-12,13,-13,14,-15,16,-17,18,-19,20,-21,22,-24,25,-26,25,-17,104,-114,196,-98,54,-126,232,-41,337,459,418,1258,2024,1306,-3081,2067,-322,1935,2599,4829,7530,3384,-1993,2649,3619,-256,-653,-863,-84,-381,-1124,-459,826,-56,-554,-375,247,362,403,-767,-54,-307,253,-528,-72,-499,-251,-538,-224,-365,-429,-387,-270,-98,-240,-189,-323,-221,-114,-200,-172,-265,-34,-233,-42,-325,-228,-392,-171,-360,-298,-341,-265,-231,-220,-299,-332,-289,-282,-307,-286,-295,-293,-265,-194,-288,-280,-341,-219,-319,-258,-367,-240,-308,-190,-294,-215,-308,-212,-271,-223,-283,-231,-230,-192,-212,-231,-187,-195,-184,-197,-212,-220,-239,-198,-224,-201,-254,-209,-229,-205,-256,-239,-265,-231,-229,-194,-202,-206,-179,-140,-169,-205,-189,-208,-194,-220,-219,-227,-219,-190,-210,-179,-230,-143,-194,-136,-216,-131,-181,-147,-210,-176,-226,-209,-222,-193,-214,-206,-190,-190,-205,-213,-200,-179,-148,-109,-85,-53,-73,-40,-100,-69,-150,-91,-144,-106,-152,-99,-85,-54,-38,-16,3,2,19,74,35,28,11,42,13,8,-18,-32,-36,-29,-19,-47,-86,-72,-66,-42,-83,-71,-83,-83,-72,-54,-20,-67,-68,-58,-74,-64,-79,-16,-60,-46,-42,-7,11,22,53,51,32,-13,33,28,3,-22,-46},
+   /* IRC_Composite_C_R0195_T195_P000.wav */
+   {12,-12,12,-12,12,-12,12,-12,12,-12,12,-12,12,-12,12,-12,11,-11,10,-9,8,-6,4,-1,-5,16,-58,-48,-46,77,37,-65,-49,-7,-27,197,332,255,848,1561,1608,-2167,729,-55,1458,1975,2964,6210,4289,99,1163,3083,300,-110,-1053,97,-72,-273,-635,580,371,-146,-5,150,225,128,22,74,-75,62,-60,-62,-162,-184,-359,-146,-99,33,-264,-53,-91,-68,-97,-68,-107,-168,-116,-102,-205,-156,-272,-180,-287,-225,-284,-228,-198,-203,-177,-240,-212,-239,-265,-245,-245,-239,-269,-232,-261,-248,-259,-262,-292,-271,-282,-251,-272,-265,-298,-265,-285,-267,-290,-263,-263,-239,-279,-219,-264,-196,-232,-191,-206,-170,-181,-179,-220,-200,-244,-184,-248,-231,-259,-234,-254,-249,-268,-279,-250,-255,-219,-236,-226,-171,-197,-163,-222,-176,-205,-187,-189,-200,-188,-195,-184,-172,-221,-174,-245,-154,-259,-156,-238,-169,-237,-202,-213,-230,-237,-242,-223,-236,-210,-202,-201,-217,-219,-206,-204,-158,-180,-128,-112,-64,-38,-58,-71,-95,-94,-116,-117,-160,-118,-117,-105,-102,-88,-47,-31,-21,-9,0,24,-11,6,-19,6,1,-16,-22,-27,-22,-10,-73,-58,-79,-51,-76,-65,-82,-99,-73,-70,-36,-82,-58,-110,-76,-105,-83,-46,-122,-54,-86,17,-33,15,14,-1,0,13,44,23,-6,2,-42},
+   /* IRC_Composite_C_R0195_T210_P000.wav */
+   {6,-6,6,-6,7,-7,7,-7,7,-7,7,-7,8,-8,8,-8,8,-9,9,-9,10,-10,10,-11,12,-12,13,-13,-3,-140,-62,16,74,-75,-81,-9,-154,52,75,214,374,933,1063,-50,-424,1215,-28,199,1671,2854,5292,2238,1086,1897,2061,-301,-116,-17,89,220,-368,-4,403,517,47,242,331,363,284,327,289,188,152,291,134,170,136,286,56,186,15,53,-103,44,-142,-179,-241,-86,-171,-94,-229,-133,-165,-58,-222,-146,-225,-113,-173,-105,-202,-191,-235,-184,-216,-202,-258,-257,-241,-211,-214,-256,-271,-278,-250,-239,-276,-254,-285,-245,-277,-265,-277,-285,-295,-284,-275,-264,-248,-242,-249,-215,-225,-190,-221,-208,-231,-195,-210,-191,-211,-203,-222,-199,-222,-228,-234,-247,-238,-233,-235,-232,-214,-216,-211,-201,-215,-188,-250,-191,-224,-183,-218,-210,-195,-212,-196,-217,-202,-221,-224,-225,-263,-219,-268,-220,-277,-225,-254,-223,-243,-232,-228,-229,-221,-217,-194,-205,-169,-170,-122,-124,-95,-89,-83,-76,-91,-74,-103,-96,-96,-80,-91,-101,-107,-123,-107,-115,-88,-113,-70,-93,-25,-70,-18,-56,-11,-38,-15,-23,-20,-23,-29,-31,-55,-70,-58,-57,-52,-93,-74,-95,-46,-80,-63,-97,-83,-108,-85,-90,-88,-102,-101,-88,-79,-54,-29,-30,-33,-9,-5,-3,24,-9,7,-23},
+   /* IRC_Composite_C_R0195_T225_P000.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-2,2,-2,2,-3,4,-4,6,-9,17,-16,-98,-60,-77,48,3,-81,-121,-43,-55,1,19,93,305,537,631,131,-260,545,567,27,1207,2128,2827,2558,1188,1325,1915,744,102,304,255,211,393,327,58,643,377,590,456,611,727,425,704,395,620,277,569,148,252,218,128,12,-134,-32,-106,-54,-94,-7,3,-19,-31,-59,-78,-49,-114,-79,-187,-72,-177,-103,-203,-112,-200,-151,-256,-183,-276,-182,-258,-181,-256,-202,-239,-241,-240,-281,-205,-284,-252,-257,-288,-259,-319,-259,-309,-239,-302,-232,-305,-231,-282,-235,-251,-227,-244,-183,-227,-177,-216,-169,-213,-156,-216,-165,-230,-183,-214,-199,-211,-207,-215,-203,-236,-236,-213,-269,-197,-284,-219,-271,-227,-256,-210,-245,-209,-247,-203,-231,-195,-235,-194,-248,-194,-245,-212,-237,-203,-241,-198,-226,-200,-216,-221,-205,-209,-210,-188,-216,-146,-202,-108,-186,-110,-157,-108,-129,-121,-125,-129,-118,-125,-104,-120,-104,-95,-101,-78,-84,-57,-71,-28,-65,-16,-83,-25,-68,-41,-63,-42,-59,-29,-46,-33,-54,-48,-58,-50,-85,-80,-93,-89,-117,-98,-103,-82,-104,-94,-118,-105,-87,-86,-85,-96,-100,-64,-69,-48,-63,-65,-45,-15,-39,-27,-55,-31},
+   /* IRC_Composite_C_R0195_T240_P000.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,-1,1,-1,1,-1,1,-1,2,-2,2,-3,5,-9,-97,-133,-196,-147,-157,-47,-93,37,-29,1,0,59,-116,-55,150,274,295,91,361,1004,1112,1011,1324,1700,2156,1559,331,574,1032,523,614,629,882,990,1087,1066,943,977,621,691,470,536,382,281,280,209,295,117,203,196,393,300,250,185,141,127,-15,-42,-136,-73,-208,-111,-261,-80,-157,-42,-115,-38,-121,-105,-155,-167,-199,-209,-235,-206,-227,-259,-253,-267,-251,-281,-214,-260,-213,-271,-234,-259,-192,-277,-204,-294,-201,-341,-226,-327,-238,-309,-206,-263,-189,-240,-185,-208,-186,-211,-192,-209,-158,-223,-155,-231,-165,-251,-159,-233,-168,-259,-189,-266,-206,-255,-200,-268,-211,-290,-218,-286,-233,-302,-227,-283,-216,-289,-207,-262,-193,-248,-196,-236,-182,-219,-173,-221,-200,-230,-207,-207,-205,-216,-209,-217,-217,-212,-207,-184,-207,-178,-189,-131,-147,-115,-129,-101,-99,-95,-93,-92,-108,-91,-105,-77,-114,-73,-104,-68,-90,-52,-74,-36,-70,-50,-80,-45,-81,-54,-78,-26,-78,-29,-94,-58,-114,-94,-144,-115,-134,-101,-99,-82,-78,-81,-65,-58,-71,-110,-125,-130,-133,-109,-102,-88,-107,-78,-90,-39,-58,-31,-64,-17,-39,-13},
+   /* IRC_Composite_C_R0195_T255_P000.wav */
+   {2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-3,3,-3,3,-3,3,-3,3,-3,4,-4,4,-5,5,-6,9,-19,-71,-126,-111,-131,-223,-213,-179,-73,-136,-101,-19,30,101,69,89,95,323,406,515,721,768,1140,1389,925,919,1504,1350,1027,1508,1800,2083,1870,1354,950,539,490,494,392,378,293,344,461,473,522,375,344,271,237,162,56,42,11,83,-5,55,-30,81,-3,71,-18,-50,0,-151,-66,-240,-143,-251,-198,-256,-250,-192,-239,-205,-271,-195,-240,-195,-267,-238,-257,-227,-242,-241,-251,-226,-256,-185,-246,-212,-252,-224,-227,-235,-216,-249,-192,-222,-170,-209,-167,-231,-184,-255,-188,-250,-181,-283,-185,-279,-146,-264,-164,-273,-161,-261,-181,-298,-216,-272,-226,-251,-181,-242,-222,-248,-216,-260,-249,-302,-271,-289,-255,-276,-256,-246,-225,-218,-198,-202,-222,-210,-223,-219,-210,-230,-209,-218,-149,-175,-141,-182,-152,-139,-146,-160,-169,-152,-140,-120,-129,-130,-102,-120,-59,-92,-67,-121,-84,-93,-62,-69,-69,-85,-101,-60,-94,-80,-123,-92,-126,-73,-83,-80,-90,-79,-71,-42,-42,-54,-59,-77,-88,-83,-86,-100,-126,-130,-119,-130,-124,-145,-127,-148,-111,-124,-92,-85,-75,-85,-70,-68,-66,-66,-59,-78,-67,-59,-65},
+   /* IRC_Composite_C_R0195_T270_P000.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,3,-4,9,-91,-159,-163,-111,-130,-163,-90,-102,-89,-163,-184,-188,-171,-83,30,36,169,298,65,214,405,559,898,896,1271,2043,2157,2404,2912,2261,1344,1103,1268,1220,756,484,263,283,379,505,398,413,455,527,338,482,334,406,255,187,211,119,189,4,182,-16,145,-56,26,-74,-83,-100,-148,-133,-197,-139,-191,-198,-217,-194,-192,-215,-224,-197,-182,-180,-242,-199,-243,-153,-261,-180,-264,-151,-250,-164,-239,-176,-235,-178,-231,-191,-258,-227,-259,-218,-280,-188,-268,-159,-259,-154,-260,-137,-244,-185,-245,-228,-251,-260,-250,-247,-239,-233,-209,-205,-224,-262,-274,-268,-248,-262,-263,-277,-234,-255,-222,-233,-226,-262,-221,-257,-211,-267,-218,-251,-186,-221,-179,-224,-195,-213,-161,-226,-193,-212,-178,-219,-191,-204,-183,-190,-175,-182,-165,-164,-140,-131,-97,-107,-80,-102,-45,-85,-60,-81,-66,-74,-59,-82,-82,-91,-88,-102,-100,-120,-85,-122,-72,-129,-98,-137,-55,-101,-62,-69,-41,-72,-56,-83,-69,-90,-53,-98,-81,-105,-80,-107,-83,-112,-102,-115,-97,-125,-111,-136,-123,-145,-118,-131,-95,-114,-62,-70,-59,-73,-48,-57},
+   /* IRC_Composite_C_R0195_T285_P000.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-3,3,-4,6,-9,19,-100,-184,-163,-233,-161,-194,-211,-190,-252,-368,-121,58,37,-12,35,111,-96,-223,-19,464,559,1425,2382,1927,1709,2204,1586,984,1062,1267,1442,1113,1078,1182,1130,1148,848,575,534,584,541,405,456,300,346,294,405,279,183,240,258,176,160,100,-27,100,-24,93,-139,0,-270,60,-268,20,-249,-123,-250,-178,-127,-185,-84,-288,-109,-237,-116,-161,-179,-137,-193,-134,-189,-164,-211,-174,-211,-244,-187,-259,-227,-263,-160,-220,-206,-234,-213,-214,-205,-225,-212,-266,-182,-254,-229,-238,-166,-272,-223,-282,-263,-305,-287,-272,-261,-288,-239,-247,-227,-247,-224,-236,-226,-259,-230,-256,-220,-217,-226,-224,-232,-214,-225,-205,-261,-251,-252,-219,-235,-230,-230,-236,-236,-209,-203,-226,-221,-222,-180,-160,-190,-161,-159,-154,-156,-97,-111,-129,-127,-107,-66,-95,-69,-92,-64,-55,-64,-43,-78,-65,-103,-71,-87,-110,-101,-114,-93,-117,-90,-112,-88,-109,-72,-82,-69,-68,-63,-65,-90,-69,-67,-77,-62,-56,-47,-91,-59,-73,-34,-101,-70,-91,-61,-105,-91,-128,-154,-149,-128,-150,-140,-142,-93,-111,-108,-78,-62,-77,-59,-62},
+   /* IRC_Composite_C_R0195_T300_P000.wav */
+   {4,-4,4,-4,4,-4,4,-4,4,-5,5,-5,5,-5,5,-5,5,-6,6,-6,6,-6,6,-7,7,-7,8,-8,8,-9,10,-11,13,-16,24,-75,-192,-243,-206,-121,-200,-413,-335,-275,-203,-85,-38,193,-95,-382,283,-596,-531,1085,630,1024,2620,2194,1518,1499,1392,607,112,846,1164,950,675,1005,1305,1174,1521,1142,1208,859,1074,771,735,552,498,679,274,454,160,457,127,176,172,280,101,64,67,-27,98,11,3,-150,-68,-72,-105,-163,-160,-139,-135,-141,-133,-136,-55,-184,-109,-200,-81,-104,-78,-213,-147,-100,-188,-136,-214,-162,-281,-139,-265,-190,-228,-184,-320,-138,-205,-130,-238,-158,-209,-100,-297,-175,-328,-233,-318,-199,-385,-256,-355,-254,-336,-269,-310,-262,-286,-237,-272,-206,-279,-217,-262,-212,-280,-260,-296,-269,-321,-249,-293,-297,-294,-263,-252,-259,-248,-231,-189,-223,-204,-205,-214,-202,-172,-186,-170,-216,-162,-160,-149,-167,-137,-119,-139,-138,-127,-91,-90,-111,-123,-105,-94,-88,-92,-101,-104,-94,-87,-104,-82,-106,-68,-106,-75,-113,-53,-76,-65,-86,-59,-75,-81,-73,-84,-59,-75,-52,-59,-73,-55,-69,-45,-87,-49,-62,-47,-76,-53,-57,-57,-74,-83,-72,-109,-94,-135,-114,-127,-112,-128,-125,-121,-104,-114,-94,-97,-102,-114,-98,-73,-61,-46,-72},
+   /* IRC_Composite_C_R0195_T315_P000.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-4,4,-4,4,-4,4,-5,5,-5,6,-7,8,-10,14,-29,-234,-230,-266,-204,-305,-247,-524,-359,-150,55,142,71,572,-570,-1004,861,-803,-374,3189,1120,2120,3144,2428,636,1026,1269,59,-187,571,1369,181,332,532,1375,1240,1554,1033,1045,866,1172,1026,1022,606,720,597,671,349,376,201,259,47,222,146,93,-124,20,-35,-174,-23,6,-91,-108,-18,6,38,-2,-118,67,-77,-79,-121,-96,-206,-171,-74,-193,-141,-167,-54,-168,-65,-180,-137,-281,-209,-164,-172,-178,-158,-127,-188,-174,-11,-152,-142,-222,-182,-348,-203,-338,-287,-380,-316,-332,-319,-347,-297,-281,-364,-362,-314,-304,-376,-347,-321,-343,-330,-367,-276,-306,-278,-290,-268,-295,-267,-229,-264,-248,-290,-244,-263,-201,-223,-197,-193,-189,-154,-159,-137,-178,-126,-217,-115,-163,-129,-166,-130,-150,-147,-107,-147,-129,-143,-117,-141,-126,-115,-135,-99,-146,-90,-112,-81,-119,-52,-80,-79,-84,-83,-64,-59,-53,-89,-91,-74,-101,-29,-121,-49,-139,-64,-94,-23,-57,-36,-29,-38,-50,-50,-47,-30,-53,-56,-81,-58,-97,-62,-69,-56,-82,-74,-103,-94,-111,-122,-150,-117,-148,-135,-117,-104,-87,-95,-77,-107,-55,-89,-56,-82,-65,-70,-36,-13},
+   /* IRC_Composite_C_R0195_T330_P000.wav */
+   {-15,15,-15,15,-15,15,-15,16,-16,16,-16,16,-16,16,-16,16,-16,16,-16,16,-16,15,-15,14,-13,11,-8,5,0,-7,12,-293,-526,-71,-505,-401,-1003,-83,-138,277,-121,450,638,-529,-2538,1974,-1021,-928,6018,1862,2117,4170,3177,-540,80,773,465,-382,760,707,287,-96,895,1260,1490,749,709,1138,1758,1405,855,490,661,833,888,382,366,310,491,203,204,155,246,122,-32,-122,-37,50,94,-114,-242,-173,94,-97,-233,-239,44,3,91,-64,-78,-19,148,9,2,-196,62,-119,-154,-277,128,-102,-103,-182,-189,-136,-49,51,-260,-285,-17,-46,-344,-260,-264,-310,-481,-356,-385,-299,-442,-338,-388,-433,-413,-352,-281,-444,-379,-419,-361,-361,-316,-320,-407,-371,-354,-283,-325,-263,-281,-264,-300,-242,-200,-218,-231,-208,-283,-185,-218,-137,-254,-173,-210,-124,-178,-189,-155,-197,-150,-148,-83,-183,-134,-157,-104,-154,-134,-133,-118,-172,-161,-120,-119,-156,-145,-96,-143,-172,-117,-71,-107,-120,-113,-53,-111,-85,-65,-51,-97,-63,-15,-77,-94,-67,-18,-45,-126,-60,-101,-21,-115,3,-114,-45,-90,-5,-69,-55,-89,-42,-99,-38,-100,-24,-95,-21,-116,-37,-113,-13,-85,-64,-143,-57,-122,-61,-126,-85,-123,-78,-94,-71,-113,-55,-72,-37,-93,-41,-42,-10,-34,-37,-13,-34},
+   /* IRC_Composite_C_R0195_T345_P000.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-2,2,-3,-196,-727,-30,-506,-136,-1125,-612,-692,-135,283,619,802,-51,-2221,1398,-964,-5308,8059,961,2013,8066,4451,-858,1438,1808,-955,-2183,475,1387,742,458,140,184,1030,1768,1316,965,1001,1155,1147,1222,1301,1124,461,289,297,592,383,273,-220,-259,-53,323,264,-32,-200,42,191,421,77,-195,-257,18,-154,-38,-140,-135,-290,-47,107,49,-29,-98,-94,-50,49,264,36,153,-148,86,-24,287,-156,101,-86,125,-186,-182,-278,-236,-501,-636,-484,-401,-330,-428,-463,-554,-283,-366,-318,-494,-441,-375,-318,-358,-465,-416,-438,-337,-400,-344,-393,-247,-308,-313,-356,-181,-229,-204,-374,-214,-320,-200,-307,-179,-339,-189,-252,-152,-246,-143,-210,-166,-196,-69,-139,-172,-216,-103,-73,-127,-214,-116,-118,-96,-247,-135,-187,-78,-264,-166,-235,-95,-173,-142,-167,-169,-129,-117,-114,-125,-125,-103,-137,-47,-110,-26,-121,-21,-72,-15,-71,-68,-34,-53,-38,-72,-96,-96,-90,-11,-126,-33,-166,-22,-92,18,-75,-35,-69,-60,-67,-82,-72,-61,-96,-68,-103,-26,-100,-19,-108,0,-110,-34,-106,-39,-54,-75,-76,-83,-35,-42,-69,-51,-59,-38,-55,19,-49,-3,-43,27,-14,-5,-46,-24}};
+
+const int16_t irc_composite_c_r0195_p015[][256] =
+  {/* IRC_Composite_C_R0195_T000_P015.wav */
+   {4,-4,4,-4,4,-4,4,-4,4,-4,4,-5,5,-5,4,-4,4,-4,3,-2,1,2,-5,-101,-665,-424,-897,-170,-507,-460,-1001,-828,-356,47,474,980,404,2103,-3700,-5968,7765,-9133,9741,10867,71,6925,3456,-1455,-2458,2718,-170,-1201,-1751,1809,507,632,-1165,1945,1684,1705,-450,1152,360,1527,269,952,644,931,727,1576,725,835,741,915,186,162,327,384,20,24,-311,60,225,484,-123,-28,-210,118,15,76,-335,-153,-249,49,-120,24,-200,-169,-213,-175,-235,-158,-280,-399,-410,-304,-104,-355,-319,-311,-79,-366,-75,-79,-120,-487,147,-42,-198,-175,135,-242,55,233,123,-314,-58,-179,-96,-352,-26,7,11,-105,-129,125,-110,-293,-276,-461,-507,-406,-422,-497,-481,-446,-459,-325,-336,-443,-420,-320,-391,-360,-309,-407,-375,-288,-289,-385,-315,-272,-292,-271,-256,-318,-173,-220,-185,-329,-142,-239,-76,-219,-117,-277,-107,-204,-117,-233,-199,-220,-177,-238,-171,-170,-187,-285,-154,-168,-145,-192,-131,-239,-194,-170,-109,-205,-155,-164,-163,-187,-126,-85,-138,-195,-140,-189,-84,-134,-22,-177,-51,-128,-26,-105,26,-144,-65,-123,-28,-136,19,-69,-43,-112,40,-43,59,-67,32,-35,97,-21,99,-27,90,23,141,21,38,39,61,12,10,-1,37,-3,7,2,41,-19,-12,-12,16,-53},
+   /* IRC_Composite_C_R0195_T015_P015.wav */
+   {100,-101,102,-102,102,-102,102,-102,101,-99,96,-93,88,-81,71,-58,37,-6,-46,154,-587,-280,-52,-712,-530,-842,-228,-637,-217,-1535,-463,274,2688,1298,-390,-2760,4349,-16532,13572,-2117,2580,22789,-850,559,1346,2618,-4202,397,316,686,-1654,1356,-848,1484,164,1873,376,1720,39,1069,309,1623,748,807,488,1192,547,1364,679,594,48,334,-179,362,98,-149,-587,-275,-81,371,-430,-372,-369,412,-126,-39,-294,169,-117,61,-209,-100,-271,-40,-262,-244,-406,12,-243,-253,-484,54,-243,-283,-575,-138,-358,-262,-307,-61,-567,-206,25,51,-696,-10,-132,-245,-388,187,-132,-194,-316,-92,-71,-291,-374,27,-295,-84,-4,145,-576,122,-217,-153,-724,53,-337,-126,-502,-305,-401,-138,-372,-296,-502,-318,-375,-140,-645,-283,-375,-217,-529,-172,-316,-324,-287,-239,-309,-305,-230,-228,-310,-332,-289,-168,-302,-286,-241,-197,-266,-181,-156,-260,-272,-185,-148,-225,-221,-239,-178,-223,-142,-203,-226,-260,-124,-148,-200,-259,-124,-203,-182,-207,-78,-186,-163,-136,-74,-179,-123,-45,-87,-180,-103,-28,-117,-135,-64,-29,-148,-90,-13,-28,-111,-40,7,-42,-98,-26,-4,-57,-22,22,-38,-72,76,67,1,-21,80,49,39,63,66,26,23,62,97,29,39,49,28,-1,52,74,-6,-4,19,15,-15,-12,17,-50},
+   /* IRC_Composite_C_R0195_T030_P015.wav */
+   {-101,102,-103,105,-105,106,-106,106,-104,102,-97,90,-79,62,-35,-12,109,-431,-649,311,-951,117,-1193,10,-1439,943,-1996,57,-1866,3892,-160,3817,-4940,7695,-14389,320,6811,-6312,29122,1548,2820,2754,2735,-5703,-20,838,313,-2387,2394,-1371,1731,262,1937,34,1456,-294,1713,221,997,660,1535,549,1179,928,928,293,321,-76,273,-164,-356,-622,24,-278,-26,-520,-116,-436,-61,-412,-21,-436,-263,-534,-25,-24,127,-328,-77,-189,107,-200,-313,-487,-278,-245,-413,-269,-216,-100,-434,-149,-238,-270,-527,-94,-144,-244,-529,-94,-120,-125,-544,-205,-477,-214,-375,-52,-404,-184,-361,-14,-189,-53,-346,-89,-276,-127,-233,-126,-417,-169,-225,-263,-470,-227,-238,-276,-329,-339,-304,-226,-209,-212,-255,-261,-275,-96,-168,-289,-298,-134,-296,-334,-291,-248,-272,-268,-362,-278,-286,-108,-360,-215,-330,-76,-394,-179,-365,-104,-449,-206,-420,-128,-409,-167,-434,-175,-416,-105,-318,-125,-408,-157,-270,-45,-277,-160,-326,-56,-220,-114,-280,-69,-199,-100,-242,-24,-143,-39,-214,3,-104,20,-184,20,-140,-7,-183,31,-135,-21,-164,13,-128,42,-188,49,-113,71,-189,90,-97,102,-87,84,-61,147,-63,149,-94,170,-108,183,-108,208,-150,179,-163,250,-98,215,-137,235,-107,214,-135,246,-126,201,-171,186,-149,135,-194,121},
+   /* IRC_Composite_C_R0195_T045_P015.wav */
+   {-146,153,-161,170,-179,189,-200,212,-224,238,-252,266,-278,280,-241,-262,-1208,369,-877,618,-1865,344,-1624,1098,-2418,1166,-2204,3338,-287,6351,-5670,4490,-9471,-5595,10415,-3658,26422,6163,571,640,3020,-3458,345,262,-17,-2379,1504,-915,2155,96,1778,-36,1408,800,1533,117,618,355,1940,474,1443,116,436,-13,320,-248,-589,-590,-422,-164,-365,-343,-251,-474,20,-343,-59,-619,-421,-460,-72,-443,-399,-435,-238,-311,-325,-256,-210,-285,-282,-222,-283,-212,-374,-178,-386,-213,-343,-35,-317,-276,-390,-79,-244,-300,-445,-222,-296,-233,-458,-263,-403,-259,-462,-188,-423,-283,-371,-78,-225,-208,-380,-203,-156,-140,-307,-202,-208,-86,-293,-179,-233,-185,-390,-305,-381,-170,-416,-344,-426,-229,-331,-201,-370,-183,-273,-88,-257,-116,-206,-60,-298,-72,-233,-115,-304,-131,-336,-187,-271,-44,-375,-190,-355,-35,-321,-189,-400,-136,-365,-198,-383,-200,-395,-244,-401,-188,-398,-184,-393,-179,-365,-150,-314,-118,-327,-118,-291,-73,-272,-87,-247,-52,-258,-27,-226,25,-210,8,-202,51,-136,62,-162,35,-135,61,-156,-12,-167,55,-165,15,-174,47,-131,28,-125,76,-91,69,-71,126,-60,104,-40,141,-51,117,-33,141,-58,103,-38,158,-48,95,-52,149,-45,86,-47,120,-68,88,-39,116,-45,74,-35,89,-68,62,-90,53},
+   /* IRC_Composite_C_R0195_T060_P015.wav */
+   {29,-34,39,-45,52,-59,69,-80,93,-110,131,-160,205,-308,-459,-920,-171,388,-674,-719,-1064,350,-1149,552,-1891,2109,-28,5773,-3032,5413,-11840,194,-1388,1646,23561,8535,2547,1107,-66,257,913,517,-131,-1162,-856,344,2136,-879,1332,-592,2048,385,1977,174,2217,-131,2091,-69,1390,-1343,963,-905,515,-1176,-81,-1557,250,-301,-119,-467,-479,-337,138,-523,-373,-525,-22,-558,-74,-857,-193,-765,-208,-1034,-15,-947,-96,-803,45,-665,15,-523,-1,-312,49,-446,-141,-311,-126,-246,-344,-467,-27,-388,-328,-427,-198,-495,-118,-356,-277,-493,-134,-525,-71,-411,-262,-445,-175,-367,-96,-400,-267,-383,-49,-313,-260,-205,-179,-166,-56,-272,-194,-146,-166,-310,-204,-405,-207,-321,-211,-468,-223,-383,-150,-326,-281,-312,-146,-220,-208,-241,-286,-189,-157,-134,-243,-213,-136,-145,-92,-188,-205,-244,-73,-253,-131,-321,-224,-277,-194,-364,-238,-342,-254,-284,-301,-272,-246,-239,-245,-226,-236,-207,-220,-191,-251,-189,-226,-188,-154,-195,-171,-163,-122,-121,-97,-111,-91,-108,-20,-120,-59,-33,-136,-42,-55,-62,-49,-97,-100,16,-97,-13,-104,7,-74,-20,-54,27,-6,45,-12,56,35,24,86,33,48,24,68,72,-3,50,22,52,12,43,-32,109,-68,106,-66,49,-61,102,-74,44,-72,28,-60,21,-71,-4,-76,16},
+   /* IRC_Composite_C_R0195_T075_P015.wav */
+   {19,-16,12,-7,2,4,-11,18,-27,36,-47,59,-68,37,-553,36,-916,-89,-13,-916,-389,-1224,572,-284,1453,853,1354,3254,1573,-10498,-1005,-5881,20609,10333,4130,9878,-3085,-1702,1001,5191,773,-2104,-1269,-41,1645,-349,309,387,447,1304,620,1533,360,1697,867,1545,-94,149,-359,473,-1422,-432,-1225,-246,-886,186,-914,-144,-892,309,-602,60,-603,-112,-520,-150,-750,-486,-681,-262,-736,-448,-1017,-103,-734,-305,-903,-61,-752,39,-644,-19,-565,33,-324,-70,-458,-205,-229,-182,-403,-322,-245,-211,-302,-350,-295,-222,-298,-293,-385,-263,-357,-235,-379,-252,-470,-139,-429,-84,-457,-178,-385,-120,-380,-232,-339,-192,-273,-205,-196,-146,-164,-174,-129,-218,-203,-145,-229,-290,-273,-262,-283,-222,-302,-215,-314,-150,-291,-151,-311,-154,-281,-177,-260,-179,-190,-177,-222,-159,-209,-215,-215,-185,-191,-224,-214,-221,-182,-209,-269,-260,-254,-234,-284,-241,-293,-201,-269,-218,-264,-188,-203,-199,-218,-207,-118,-151,-146,-184,-126,-92,-92,-129,-108,-109,-42,-120,-57,-144,-45,-102,-23,-164,-63,-98,-36,-106,-81,-3,-74,-55,-79,-49,-60,-59,-60,-31,-36,-35,57,-37,99,-7,135,-26,115,14,154,-6,78,28,119,-3,53,-48,80,-38,55,-56,13,-65,74,-45,19,-79,50,-39,24,-95,40,-41,15,-61,55,-129},
+   /* IRC_Composite_C_R0195_T090_P015.wav */
+   {106,-108,111,-113,116,-119,121,-123,125,-125,121,-102,-150,-386,-21,-846,-117,203,-762,-759,-425,-20,26,535,183,3678,866,180,877,-12284,4742,-888,21444,10158,-2546,4612,-145,-630,3778,3819,1940,-4361,-1372,2229,1116,-1596,-423,1713,840,536,1040,2123,618,888,-302,898,-618,52,-845,240,-1279,-435,-1143,105,-929,-50,-1077,-180,-541,98,-930,-161,-671,-158,-656,-338,-774,-263,-620,-260,-792,-238,-787,-172,-874,-235,-797,-199,-767,-178,-702,-101,-607,-63,-479,-72,-504,-49,-400,-118,-387,-113,-404,-73,-442,-100,-429,-15,-496,-79,-480,-58,-494,-80,-498,-112,-411,-138,-447,-136,-406,-145,-485,-166,-440,-131,-485,-87,-417,-104,-413,-102,-426,-31,-350,-62,-299,27,-272,-36,-302,-22,-320,-120,-364,-124,-307,-111,-382,-185,-231,-108,-275,-163,-225,-123,-157,-158,-254,-183,-208,-146,-274,-183,-229,-148,-301,-148,-275,-212,-388,-177,-304,-193,-358,-180,-265,-195,-243,-171,-184,-212,-200,-212,-156,-173,-149,-232,-145,-162,-100,-147,-119,-102,-57,-53,-87,-18,-36,-11,-87,-46,-46,17,-89,-82,-57,6,-100,-61,-101,-38,-60,-46,-108,-40,-41,19,-104,-21,-14,77,-24,51,22,136,25,103,19,102,3,92,-42,26,-51,25,-38,-15,-65,-12,-35,-8,28,24,-31,-1,36,92,2,6,8,52,-33,-7,-25,-27},
+   /* IRC_Composite_C_R0195_T105_P015.wav */
+   {16,-19,23,-28,34,-40,48,-57,68,-82,100,-121,33,-823,199,-561,-96,-481,741,-1752,-67,-480,1840,-966,591,1601,5280,-3652,770,-7901,214,10553,2446,18816,1377,-3359,-1411,6633,5595,-115,-232,1823,-2575,-717,-242,2719,-2255,509,511,2747,225,1585,549,951,-478,-34,-731,-13,-1246,-512,-881,-341,-1127,207,-442,-495,-702,-517,-566,35,-620,-983,-450,-179,-571,-441,-658,-297,-211,-340,-730,-514,-334,-479,-562,-662,-463,-524,-336,-537,-522,-336,-226,-483,-343,-332,-364,-181,-306,-365,-271,-175,-316,-213,-283,-326,-115,-221,-216,-290,-146,-290,-98,-334,-301,-279,-138,-349,-278,-318,-250,-285,-237,-388,-326,-257,-206,-416,-256,-293,-207,-350,-156,-397,-149,-314,-125,-323,-153,-327,-64,-258,-145,-241,-10,-249,-51,-197,-108,-234,33,-286,-138,-267,-26,-273,-117,-347,-77,-229,-117,-301,-119,-229,-104,-247,-175,-311,-162,-227,-248,-287,-181,-293,-170,-318,-206,-285,-159,-334,-167,-232,-161,-235,-76,-246,-154,-204,-10,-221,-106,-229,14,-182,-12,-251,30,-154,32,-190,61,-152,90,-145,48,-122,71,-96,29,-121,6,-57,66,-80,40,-49,59,-112,-8,-129,54,-18,76,-21,97,37,171,23,59,-61,75,-37,21,-127,1,-154,68,-112,-12,-180,86,-58,91,-120,128,-28,129,-100,167,-17,106,-84,94,-49,76,-37},
+   /* IRC_Composite_C_R0195_T120_P015.wav */
+   {39,-39,39,-39,39,-38,38,-37,35,-32,27,-19,0,-223,40,-675,329,-841,457,-348,-274,-1647,1770,432,-494,3,4260,289,887,-4454,-3947,7446,-2378,15614,7773,95,-2028,2409,7171,2953,950,1439,-1677,-1118,-1770,1575,491,137,-1226,1181,2451,1325,-149,862,100,-196,-613,-302,-1148,-1325,-632,-731,-1011,-688,174,-67,-430,-378,-168,-383,-445,-570,-934,-576,-71,-333,-939,-576,-180,-299,-551,-542,-536,-312,-421,-506,-398,-463,-360,-423,-342,-463,-390,-485,-299,-416,-291,-519,-227,-313,-272,-376,-199,-241,-176,-354,-259,-245,-94,-289,-288,-266,-79,-203,-210,-332,-200,-224,-227,-331,-258,-315,-219,-293,-232,-358,-231,-306,-201,-298,-234,-413,-216,-295,-220,-417,-228,-272,-126,-326,-197,-254,-104,-275,-161,-280,-75,-215,-110,-276,-33,-130,-19,-261,-8,-171,30,-242,-55,-229,-65,-254,-74,-218,-134,-301,-99,-211,-155,-300,-156,-234,-131,-322,-164,-311,-142,-349,-229,-334,-148,-297,-165,-289,-133,-215,-100,-254,-101,-185,-51,-234,-38,-186,-6,-168,0,-111,-29,-125,29,-96,-23,-153,22,-59,-32,-103,28,-118,11,-64,83,-103,41,-112,28,-109,99,-41,39,-86,157,43,121,-15,157,-11,121,-30,90,-113,25,-109,38,-172,-47,-151,15,-102,21,-97,16,-33,104,12,59,-55,121,-5,63,-116,114,-30,115,-85},
+   /* IRC_Composite_C_R0195_T135_P015.wav */
+   {15,-17,19,-21,24,-28,32,-37,43,-50,60,-73,90,-115,157,-297,361,-673,287,-116,-598,-544,1021,-902,-263,1113,1563,-848,3395,810,-1816,-1445,-4755,10984,3295,5362,6262,2589,-529,1444,7698,4758,-2233,-2138,-1622,980,-226,100,-517,871,952,682,983,1138,-542,-341,-164,-377,-526,-992,-910,-828,-413,-830,-380,-392,-124,-316,-225,-315,-25,-678,-691,-468,-412,-545,-395,-676,-301,-467,-356,-548,-300,-516,-260,-553,-295,-485,-232,-525,-220,-421,-217,-488,-227,-455,-268,-426,-269,-474,-232,-395,-199,-378,-219,-372,-176,-401,-202,-354,-114,-340,-124,-267,-86,-322,-166,-298,-181,-316,-179,-307,-163,-290,-175,-321,-186,-325,-234,-372,-200,-335,-202,-352,-199,-268,-180,-319,-236,-296,-200,-331,-209,-300,-142,-273,-118,-226,-91,-203,-97,-250,-91,-176,-72,-201,-65,-156,-22,-171,-59,-200,-36,-210,-93,-201,-48,-197,-73,-185,-73,-195,-95,-266,-166,-294,-169,-299,-216,-315,-187,-265,-167,-287,-146,-275,-175,-265,-144,-282,-127,-249,-66,-222,-39,-146,-32,-126,-20,-76,5,-92,18,-56,21,-91,13,-119,-23,-127,37,-71,16,-116,10,-105,12,-96,55,-69,43,-76,137,-20,86,-3,123,28,100,8,49,-58,60,-31,-10,-117,-15,-44,-19,-89,2,-51,-18,-51,62,-39,31,-46,59,-59,67,-57,112,-72,62,-22},
+   /* IRC_Composite_C_R0195_T150_P015.wav */
+   {22,-23,24,-24,25,-26,27,-27,28,-28,28,-28,26,-24,19,-9,-13,134,-205,5,10,-146,-122,-75,-78,-152,345,-100,1711,198,1882,367,2730,-6333,870,5933,2367,6468,4276,4587,970,719,5822,3692,-1207,-2971,-1058,797,482,-202,-740,507,536,875,480,668,-204,-211,-690,500,-1107,-874,-841,-223,-910,-122,-465,-407,-494,-192,-244,-202,-337,-294,-451,-519,-512,-353,-580,-345,-407,-307,-608,-176,-450,-303,-475,-205,-453,-277,-342,-237,-284,-212,-314,-272,-380,-283,-368,-307,-417,-297,-337,-321,-374,-329,-283,-379,-283,-310,-317,-249,-232,-217,-269,-190,-252,-183,-270,-238,-227,-187,-286,-191,-260,-230,-245,-233,-289,-230,-246,-218,-241,-208,-260,-204,-276,-211,-286,-243,-284,-211,-281,-227,-258,-199,-238,-185,-241,-133,-220,-133,-189,-80,-190,-65,-146,-88,-167,-80,-129,-110,-166,-124,-142,-139,-178,-111,-146,-116,-143,-102,-118,-138,-137,-152,-173,-190,-174,-190,-193,-230,-208,-208,-216,-246,-203,-231,-209,-216,-167,-210,-170,-158,-128,-148,-142,-127,-104,-88,-98,-82,-75,-58,-53,-46,-85,-53,-56,-1,-14,17,-80,-58,-41,-57,-6,-70,6,-78,37,-14,58,10,69,35,20,45,42,27,3,3,-29,-5,-10,26,-54,-21,-21,39,-58,-58,-23,10,-16,-35,6,-22,2,-8,54,-13,7,-2,51,6},
+   /* IRC_Composite_C_R0195_T165_P015.wav */
+   {20,-21,21,-22,23,-23,24,-25,26,-26,27,-28,29,-29,30,-30,29,-28,22,-5,163,-209,156,-276,329,-325,157,-151,120,93,1138,522,872,2060,255,503,-4749,3937,3841,3248,5654,4851,3110,1024,1230,4380,981,-2433,-2102,345,461,292,-696,50,937,951,-195,227,100,-111,-682,-242,-295,-695,-542,-371,-167,-471,-237,-441,-313,-257,-355,-428,-338,-330,-347,-256,-340,-455,-298,-395,-419,-365,-319,-336,-313,-224,-306,-223,-245,-277,-258,-289,-268,-318,-289,-331,-299,-363,-319,-390,-363,-384,-327,-342,-339,-372,-297,-284,-259,-319,-264,-296,-233,-297,-258,-289,-255,-235,-239,-231,-238,-221,-219,-255,-237,-261,-205,-265,-219,-253,-180,-250,-196,-250,-200,-248,-207,-278,-194,-276,-219,-261,-199,-283,-219,-241,-187,-217,-180,-196,-141,-180,-121,-175,-110,-181,-79,-171,-84,-195,-62,-164,-99,-185,-115,-180,-164,-155,-140,-152,-147,-164,-113,-154,-117,-178,-132,-201,-137,-183,-162,-227,-171,-200,-151,-199,-165,-161,-150,-178,-144,-157,-158,-179,-144,-176,-163,-168,-98,-168,-122,-152,-52,-148,-86,-125,-9,-56,-37,-107,-79,-44,-26,-30,-92,-55,-14,29,14,24,62,67,48,30,48,10,57,-9,40,-55,25,-77,17,-96,25,-64,21,-101,-1,4,58,-21,25,24,70,3,30,9,36,-16,6,-21,-39,-15},
+   /* IRC_Composite_C_R0195_T180_P015.wav */
+   {-9,9,-9,9,-10,10,-10,10,-11,11,-11,11,-12,12,-13,13,-14,15,-15,16,-17,19,-21,50,-66,25,121,28,-120,155,-168,335,416,671,222,2210,806,1236,-2882,612,2593,898,4860,6139,4465,870,-196,3886,2840,-2197,-1283,-67,115,-435,2,221,731,-37,-52,179,744,-513,-450,-510,328,-437,-225,-503,1,-204,-76,-530,-331,-379,-166,-435,-279,-284,-151,-266,-225,-384,-293,-305,-154,-293,-219,-273,-134,-244,-264,-371,-311,-317,-245,-345,-336,-343,-237,-329,-370,-441,-342,-336,-321,-387,-336,-319,-260,-281,-304,-299,-268,-249,-331,-318,-306,-253,-254,-267,-280,-263,-244,-262,-278,-297,-251,-213,-235,-246,-260,-197,-223,-206,-272,-188,-228,-176,-268,-213,-279,-202,-229,-233,-258,-216,-166,-186,-223,-208,-202,-136,-198,-161,-225,-147,-174,-123,-192,-165,-169,-119,-164,-157,-169,-110,-156,-134,-190,-131,-196,-117,-175,-127,-176,-95,-111,-96,-149,-112,-153,-144,-212,-161,-218,-171,-217,-170,-191,-155,-154,-114,-156,-141,-134,-77,-145,-152,-168,-124,-167,-164,-158,-156,-173,-143,-139,-75,-112,-48,-151,-85,-139,-30,-81,-70,-106,-105,-40,-41,29,15,57,54,103,128,65,65,35,45,-4,-23,-56,-53,-39,-44,-11,-66,11,-23,53,-3,44,26,45,23,36,66,32,13,3,39,11,-14,8,-12},
+   /* IRC_Composite_C_R0195_T195_P015.wav */
+   {3,-3,4,-4,4,-5,5,-5,6,-6,7,-7,8,-9,9,-10,11,-13,14,-16,19,-22,27,-33,44,-87,114,-132,102,-154,155,-97,-20,-37,-29,363,412,474,853,1336,1092,-788,-1724,2098,1099,2640,4844,4905,2895,-39,1681,2632,175,-1400,-248,325,22,-206,196,419,561,41,192,277,401,-173,2,-114,106,-178,-24,-176,30,-91,54,-173,-183,-124,-65,-136,-127,-106,-163,-59,-115,-115,-267,-191,-291,-145,-314,-229,-397,-205,-300,-229,-373,-215,-360,-218,-346,-234,-397,-220,-360,-257,-382,-254,-389,-213,-322,-234,-324,-214,-345,-227,-381,-255,-363,-174,-354,-207,-318,-230,-299,-263,-316,-271,-243,-219,-239,-195,-263,-183,-262,-181,-310,-210,-258,-227,-229,-243,-215,-263,-194,-269,-182,-235,-184,-209,-147,-196,-148,-196,-147,-207,-155,-221,-187,-223,-184,-221,-208,-205,-186,-173,-165,-148,-132,-141,-127,-151,-120,-154,-131,-136,-104,-119,-115,-117,-108,-145,-140,-188,-146,-212,-174,-199,-183,-203,-176,-190,-174,-178,-155,-183,-156,-192,-139,-178,-158,-185,-162,-172,-198,-158,-136,-126,-116,-127,-125,-157,-109,-154,-85,-152,-60,-135,-26,-117,10,-26,16,-15,36,69,64,91,41,52,-5,31,-40,-16,-60,-3,-25,-20,-13,18,23,6,15,15,42,21,45,37,32,17,30,46,26,13,23,-19},
+   /* IRC_Composite_C_R0195_T210_P015.wav */
+   {5,-5,6,-6,6,-6,6,-6,6,-6,7,-7,7,-7,7,-8,8,-8,8,-9,9,-9,9,-9,9,-9,7,-1,-45,-76,21,-111,31,-35,-9,-141,17,-81,104,351,414,314,1167,1284,-698,-334,-392,1578,2039,2332,5286,3230,840,735,2013,886,-250,-668,388,293,-154,194,485,539,373,247,362,554,379,14,200,206,243,111,161,197,127,193,183,265,-4,123,-60,64,-188,-26,-316,-89,-293,-150,-287,-153,-256,-136,-244,-221,-246,-144,-274,-195,-276,-224,-307,-223,-340,-289,-302,-287,-323,-301,-317,-303,-243,-292,-298,-294,-312,-280,-306,-243,-309,-226,-276,-214,-262,-251,-287,-252,-261,-239,-268,-241,-251,-220,-263,-226,-289,-222,-266,-235,-299,-243,-268,-249,-232,-227,-224,-212,-188,-201,-205,-192,-215,-147,-197,-162,-188,-158,-182,-181,-170,-210,-185,-204,-179,-201,-174,-187,-176,-157,-173,-157,-158,-147,-138,-131,-123,-128,-126,-106,-147,-126,-173,-126,-184,-147,-199,-168,-203,-173,-188,-192,-200,-201,-192,-187,-199,-212,-196,-185,-179,-192,-178,-197,-173,-188,-148,-169,-132,-157,-149,-165,-140,-138,-136,-123,-135,-83,-122,-54,-79,-22,-60,-30,-15,-6,29,-8,28,-22,45,-13,39,-33,30,5,20,-21,0,10,30,-5,10,-9,35,17,55,16,50,34,50,43,21,55,18,41},
+   /* IRC_Composite_C_R0195_T225_P015.wav */
+   {3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-2,2,-2,2,-1,1,0,-5,-60,-132,-64,-20,-85,-7,-176,-5,-52,-74,-72,84,430,320,474,759,338,-353,-244,418,1855,1977,2351,3573,2234,1049,638,1145,830,98,-47,162,541,334,496,426,623,595,701,586,611,652,468,517,289,478,320,343,117,277,121,176,-24,22,-90,-30,-74,-66,-98,-115,-117,-110,-81,-130,-60,-133,-100,-189,-113,-212,-176,-266,-224,-323,-264,-344,-286,-346,-280,-345,-296,-322,-294,-296,-283,-287,-282,-269,-247,-271,-247,-275,-227,-292,-244,-283,-265,-275,-259,-269,-255,-263,-256,-260,-233,-264,-234,-273,-263,-267,-260,-261,-252,-268,-235,-241,-202,-245,-183,-233,-168,-240,-165,-221,-162,-203,-155,-192,-162,-177,-157,-183,-162,-190,-153,-205,-157,-215,-152,-221,-147,-199,-142,-182,-144,-171,-142,-156,-148,-173,-154,-163,-157,-167,-170,-162,-167,-179,-174,-184,-178,-180,-174,-206,-178,-198,-188,-207,-204,-201,-199,-190,-214,-206,-229,-200,-212,-173,-192,-170,-215,-166,-189,-131,-174,-137,-168,-105,-105,-64,-83,-43,-48,2,-25,-6,-28,14,-21,8,-16,16,3,33,17,37,28,49,34,62,28,60,13,34,6,35,-8,13,-2,6,11,8,6,-3,12},
+   /* IRC_Composite_C_R0195_T240_P015.wav */
+   {0,0,0,0,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,3,-3,3,-4,4,-5,6,-7,9,-12,19,-47,-65,-42,-103,-56,-86,-61,-84,-136,-84,-125,-76,46,139,205,243,514,311,-296,-329,431,1034,1553,1739,2266,2575,1097,709,1340,1199,680,407,592,719,747,950,891,924,905,835,742,679,646,246,443,284,396,76,270,79,360,141,234,132,176,91,44,66,-33,37,-24,-26,-52,-140,-134,-171,-129,-209,-205,-236,-225,-189,-236,-238,-284,-228,-319,-232,-313,-237,-314,-261,-304,-270,-301,-338,-291,-284,-273,-275,-283,-246,-253,-222,-299,-253,-305,-231,-286,-251,-299,-224,-259,-244,-287,-247,-273,-229,-277,-233,-278,-222,-256,-210,-244,-213,-226,-197,-200,-180,-187,-177,-191,-166,-184,-157,-198,-181,-196,-166,-179,-163,-184,-173,-192,-155,-187,-160,-198,-161,-183,-151,-173,-150,-191,-183,-210,-175,-205,-187,-208,-163,-185,-168,-170,-162,-180,-159,-180,-156,-158,-160,-186,-181,-225,-214,-217,-218,-234,-238,-231,-223,-221,-238,-230,-217,-206,-178,-170,-207,-185,-183,-154,-152,-154,-148,-132,-105,-76,-64,-31,-49,-9,-20,6,-12,13,19,25,27,32,34,31,31,36,25,9,26,16,20,-7,0,-5,22,12,4,3,18,14,9,-1,-12,-43},
+   /* IRC_Composite_C_R0195_T255_P015.wav */
+   {1,-1,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,3,-3,3,-3,3,-4,5,-6,10,-30,-120,-103,-93,-24,-84,-108,-106,-140,-67,-76,-73,-121,-133,-119,68,29,241,313,294,61,72,517,913,1238,1676,2099,2348,2019,1198,861,1666,1655,972,1040,933,753,835,866,579,456,471,608,472,520,250,352,304,298,249,242,223,199,145,120,235,93,122,49,3,-39,-15,-81,-71,-213,-81,-243,-130,-266,-181,-248,-213,-210,-258,-180,-292,-166,-311,-258,-301,-258,-288,-304,-246,-283,-255,-296,-284,-260,-336,-252,-351,-234,-370,-214,-328,-219,-303,-234,-296,-242,-261,-235,-272,-252,-263,-194,-230,-174,-222,-179,-253,-165,-264,-178,-236,-165,-217,-158,-210,-155,-187,-164,-190,-169,-218,-178,-196,-162,-199,-188,-233,-170,-224,-161,-189,-146,-207,-166,-198,-168,-188,-182,-169,-203,-189,-199,-180,-192,-203,-183,-200,-212,-208,-215,-216,-240,-188,-209,-178,-223,-162,-198,-167,-209,-211,-239,-196,-203,-168,-214,-197,-239,-145,-209,-149,-231,-167,-214,-131,-185,-129,-161,-129,-137,-105,-119,-97,-124,-84,-109,-56,-30,29,13,51,35,66,39,42,23,8,-3,12,8,29,13,32,3,26,10,21,-16,-7,-1,7,2,-5,-3,-3,-1},
+   /* IRC_Composite_C_R0195_T270_P015.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-2,2,-2,3,-5,38,41,81,155,185,109,43,168,195,276,233,321,381,601,818,813,1005,694,582,706,813,1380,2408,2859,3417,3192,1234,164,601,379,145,-151,-609,-543,-195,49,-120,-111,136,61,-67,-105,-152,-114,-346,-267,-397,-310,-240,-454,-334,-493,-338,-595,-418,-579,-472,-506,-514,-489,-592,-416,-592,-471,-574,-491,-531,-515,-529,-526,-514,-446,-512,-519,-497,-434,-454,-430,-467,-415,-434,-400,-422,-405,-387,-422,-391,-428,-343,-360,-326,-389,-310,-335,-241,-318,-246,-341,-202,-264,-207,-278,-221,-234,-239,-210,-224,-186,-240,-226,-220,-207,-184,-222,-220,-245,-206,-189,-231,-234,-238,-195,-237,-129,-227,-188,-216,-145,-224,-225,-241,-252,-274,-214,-216,-202,-212,-164,-190,-171,-187,-134,-175,-141,-168,-164,-167,-150,-174,-138,-128,-101,-135,-76,-112,-70,-116,-70,-102,-92,-120,-98,-123,-112,-146,-92,-145,-88,-135,-96,-115,-84,-87,-67,-61,-82,-42,-48,11,-13,5,25,35,52,70,95,98,101,116,108,116,94,137,109,102,83,89,88,89,64,59,40,43,14,53,3,16,-3,7,12,-13,16,-2,28,6,37,18},
+   /* IRC_Composite_C_R0195_T285_P015.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,2,-3,6,-109,-159,-182,-252,-225,-252,-209,-220,-323,-249,-123,27,108,237,39,-227,-176,-141,26,983,1563,1989,2256,2042,1443,1643,1525,1088,1602,1474,1170,1150,824,708,924,1176,1036,269,443,781,769,425,308,405,367,433,461,448,234,194,163,218,246,100,-62,56,-52,109,-114,68,-271,33,-231,-10,-251,-107,-211,-189,-193,-271,-173,-258,-220,-344,-246,-222,-245,-241,-310,-276,-282,-207,-275,-244,-325,-317,-257,-224,-252,-269,-249,-255,-262,-227,-233,-233,-271,-256,-189,-188,-164,-249,-173,-198,-150,-210,-124,-172,-127,-154,-156,-239,-193,-248,-200,-220,-169,-212,-123,-176,-157,-249,-249,-297,-272,-295,-303,-316,-296,-263,-235,-217,-173,-234,-208,-250,-182,-255,-227,-234,-247,-244,-205,-187,-219,-199,-160,-156,-146,-178,-163,-199,-166,-183,-164,-214,-208,-196,-160,-168,-196,-193,-160,-204,-162,-188,-175,-206,-218,-198,-177,-203,-194,-169,-169,-172,-168,-160,-149,-149,-113,-128,-93,-92,-23,-69,-13,-56,-10,-44,8,-28,9,-21,0,13,41,16,24,3,43,28,54,11,12,-13,29,-17,4,-27,25,-15,29,-5,31,21,23},
+   /* IRC_Composite_C_R0195_T300_P015.wav */
+   {3,-3,3,-3,3,-3,3,-3,4,-4,4,-4,4,-4,4,-4,4,-4,5,-5,5,-5,5,-6,6,-6,6,-7,7,-8,9,-10,13,-17,28,-93,-234,-180,-207,-251,-293,-262,-250,-346,-376,-130,67,-6,108,-151,-710,-124,223,55,1941,2161,1520,2079,1404,552,750,680,497,384,1277,1177,733,896,1394,1657,1268,1283,1309,1208,892,805,954,655,670,296,482,320,570,596,507,445,118,348,106,469,16,25,-157,191,5,47,-157,-78,-270,-124,-129,-61,-219,-206,-188,-118,-157,-193,-216,-244,-222,-155,-190,-248,-263,-189,-260,-255,-305,-285,-272,-179,-250,-320,-353,-177,-207,-158,-311,-279,-294,-136,-181,-176,-231,-183,-184,-86,-72,-138,-172,-154,-185,-208,-132,-194,-245,-197,-155,-169,-170,-257,-250,-273,-291,-308,-267,-292,-322,-261,-262,-223,-282,-280,-302,-284,-249,-260,-293,-276,-203,-199,-209,-219,-251,-230,-225,-236,-239,-254,-234,-253,-179,-216,-170,-211,-150,-212,-180,-190,-146,-198,-192,-173,-159,-174,-166,-187,-188,-211,-146,-166,-163,-201,-184,-171,-129,-130,-154,-144,-145,-167,-132,-161,-149,-160,-132,-127,-113,-104,-89,-73,-95,-76,-83,-55,-53,-22,-5,-13,-3,5,22,0,15,5,-11,-15,15,-13,-5,-8,7,-15,26,37,33,12,52,50,45,34,80,35,79,31},
+   /* IRC_Composite_C_R0195_T315_P015.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,2,-2,2,-3,-145,-419,-273,-234,-320,-494,-368,-332,-137,23,435,141,-274,189,-707,-1152,1545,415,1656,4094,1604,1580,2053,473,-525,351,1046,431,4,613,555,843,494,1071,1048,1375,1139,994,1431,1283,1617,838,1011,491,1146,716,701,358,388,443,310,457,32,-5,-93,169,269,-27,91,-165,86,-189,135,-120,-103,-270,-74,-56,-217,-55,-238,-119,-288,-120,-263,-264,-223,-279,-179,-302,-173,-211,-262,-189,-255,-177,-382,-172,-263,-206,-302,-155,-222,-165,-156,-91,-226,-129,-208,-104,-116,-7,-182,-57,-43,-59,-165,-192,-254,-334,-329,-222,-307,-218,-299,-183,-230,-153,-307,-262,-341,-352,-360,-306,-320,-349,-329,-284,-313,-262,-287,-276,-348,-332,-282,-281,-235,-311,-213,-232,-205,-232,-209,-189,-252,-237,-224,-214,-199,-229,-157,-205,-196,-185,-173,-156,-212,-157,-188,-156,-172,-131,-151,-175,-157,-139,-172,-126,-151,-157,-177,-123,-115,-119,-116,-103,-151,-157,-157,-103,-111,-83,-143,-134,-106,-78,-57,-83,-76,-143,-96,-110,-30,-81,-37,-72,-10,-21,-9,18,1,-1,-42,21,2,14,26,52,25,26,34,53,32,44,40,74,42,66,49,90,62,69,50},
+   /* IRC_Composite_C_R0195_T330_P015.wav */
+   {1,-1,1,0,0,0,0,0,0,0,0,1,-1,1,-2,2,-2,3,-3,4,-5,6,-7,9,-11,14,-18,25,-38,72,-494,-377,-163,-458,-490,-5,-649,-505,-448,737,42,720,-5,-1466,655,-1957,-272,3638,1401,3790,5414,320,1005,1351,716,-1658,736,755,629,-241,367,775,655,1313,522,711,607,1017,814,1153,1014,727,1072,1193,1369,911,877,665,747,788,602,389,-2,301,79,299,235,78,-171,-171,-12,-204,-96,-124,-285,-390,-109,136,-127,-79,-244,-125,-236,-21,-257,-313,-264,-266,-248,-188,-112,-345,-304,-50,-265,-235,-235,-38,-345,-176,-119,-52,-156,-49,-98,-184,-269,-46,-139,-85,-206,-15,-64,-75,-25,28,-69,-306,-216,-145,-321,-127,-396,-359,-364,-161,-402,-370,-313,-329,-419,-337,-409,-317,-386,-319,-505,-396,-337,-244,-368,-365,-380,-286,-289,-230,-330,-297,-331,-175,-233,-184,-243,-195,-204,-144,-176,-187,-238,-158,-239,-170,-186,-124,-214,-122,-133,-166,-196,-122,-151,-204,-200,-147,-209,-159,-148,-140,-151,-108,-106,-148,-137,-107,-109,-157,-138,-152,-159,-152,-143,-132,-169,-84,-121,-99,-110,-53,-57,-49,-65,-56,-59,-72,-39,-80,-76,-50,-24,-78,-31,-4,10,-54,43,12,18,43,54,13,-6,86,38,22,50,62,50,38,99,31,62,60,58,29,11,38},
+   /* IRC_Composite_C_R0195_T345_P015.wav */
+   {-11,11,-12,12,-13,13,-14,15,-16,16,-17,18,-19,20,-22,23,-24,26,-27,29,-32,34,-38,43,-52,76,-328,-363,-214,-611,-509,-335,-643,-447,-869,31,57,1040,355,-374,1188,-2297,-4332,5821,-1811,4636,8422,1800,2573,2446,333,-2543,348,441,1038,-1221,-120,503,1705,557,994,621,1108,529,1387,553,713,268,987,1168,1373,742,1063,633,1079,800,733,478,430,-8,448,243,389,-34,240,-156,40,-68,208,-262,-163,-284,-85,-119,4,-292,-169,-173,-9,-177,-185,-284,-261,-239,-151,-333,-444,-278,-212,-269,-320,-207,-353,-167,-122,5,-101,-146,-236,26,158,-193,-6,31,-110,-13,194,-336,-291,166,-187,-339,71,83,-256,-36,64,-291,-253,-258,-423,-483,-404,-542,-450,-398,-404,-517,-415,-395,-300,-370,-455,-406,-370,-334,-365,-318,-452,-367,-267,-228,-288,-345,-283,-288,-195,-237,-211,-198,-242,-243,-148,-120,-215,-182,-175,-171,-178,-118,-143,-239,-153,-169,-160,-247,-180,-208,-203,-145,-188,-199,-188,-106,-162,-137,-157,-131,-109,-183,-154,-130,-109,-197,-150,-64,-168,-155,-207,-78,-176,15,-176,-113,-192,-34,-58,-13,-144,-90,-50,-81,-89,24,-35,-100,-56,-23,-9,29,37,-35,-51,58,55,55,-8,24,51,71,28,39,46,49,51,27,-11,48,41,22,-3,23,-14,20,-19,-13}};
+
+const int16_t irc_composite_c_r0195_p030[][256] =
+  {/* IRC_Composite_C_R0195_T000_P030.wav */
+   {-54,55,-55,56,-57,57,-58,58,-58,59,-58,58,-58,56,-55,52,-48,42,-34,21,1,-38,103,-854,-293,-223,-413,-662,-699,-525,-649,-153,-821,-339,218,2022,2054,-2330,-2792,1747,-9777,10764,1443,5425,12121,258,-1458,190,2417,73,-781,460,-747,-560,1297,150,875,435,1456,292,1300,-209,948,616,962,-100,649,571,794,577,979,597,904,611,978,804,976,113,890,241,591,176,615,-381,92,-14,208,-196,144,-25,-87,-290,-51,-201,-11,-376,13,-401,6,-326,16,-348,-73,-269,-196,-276,-77,-266,-298,-332,-42,-326,-208,-519,-147,-423,-93,-432,-257,-451,-109,-270,-192,-375,-151,-315,-188,-206,-37,-459,-168,-241,-10,-278,-73,-257,87,-155,-18,-234,-126,-305,-82,-294,-405,-294,-172,-409,-47,-89,-22,-89,128,-65,30,48,-38,-108,-174,-371,-254,-219,-255,-366,-375,-485,-192,-333,-204,-374,-171,-415,-254,-437,-213,-353,-257,-320,-265,-229,-230,-199,-344,-342,-208,-291,-328,-415,-224,-366,-172,-311,-156,-289,-178,-259,-84,-212,-136,-195,-136,-204,-23,-192,-135,-219,-38,-190,-41,-200,-34,-195,-41,-117,-47,-160,-96,-111,-97,-131,-73,-186,-78,-137,-70,-202,-93,-119,-55,-149,-67,-139,-48,-119,-34,-122,-39,-119,-31,-62,-1,-61,-28,-48,31,6,17,-8,75,29,40,45,65,21,45,14},
+   /* IRC_Composite_C_R0195_T015_P030.wav */
+   {45,-49,52,-57,61,-66,71,-77,83,-90,98,-107,116,-126,138,-150,164,-177,188,-179,-73,-966,-102,-989,560,-1173,-327,-1684,597,-1115,259,-1628,2183,1183,3440,-5302,3131,-10502,-529,12738,-6827,21598,5258,-7,-3187,1963,495,327,502,-665,-1769,1964,-371,1009,596,896,151,1819,-137,401,915,974,249,114,559,962,751,584,371,576,838,1123,906,351,-23,332,521,427,-138,-265,-100,55,-54,59,-205,-261,-77,2,-107,-229,-219,-403,-235,-134,-143,-470,-234,-177,-61,-236,-83,-258,-197,-192,-101,-300,-331,-254,-216,-242,-271,-289,-310,-261,-262,-333,-321,-293,-270,-333,-255,-243,-275,-316,-256,-252,-176,-243,-182,-212,-172,-289,-118,-228,-303,-307,-77,-225,-298,-230,-122,-403,-282,-152,-387,-451,8,13,-160,-49,124,-112,-104,87,-45,-358,-217,-135,-206,-167,-172,-229,-317,-197,-162,-307,-272,-363,-290,-325,-168,-360,-289,-359,-212,-286,-239,-278,-241,-270,-234,-333,-266,-308,-234,-344,-218,-357,-274,-331,-208,-244,-205,-284,-219,-226,-149,-177,-181,-201,-159,-198,-109,-172,-113,-218,-95,-145,-67,-151,-73,-148,-84,-129,-52,-118,-51,-142,-54,-113,-29,-150,-66,-118,-44,-150,-8,-105,-59,-133,-10,-103,-51,-102,-26,-114,2,-62,-14,-78,24,-62,35,-36,28,-33,74,-22,57,13,101,-23,94,36,50,32},
+   /* IRC_Composite_C_R0195_T030_P030.wav */
+   {22,-22,22,-22,22,-22,21,-21,20,-19,18,-16,13,-9,3,6,-23,62,-927,-68,-994,540,-922,-246,-1369,328,-1205,-194,-1442,1439,570,2758,-885,3393,-4925,-8851,7444,-11725,26992,7407,2376,4575,-342,-2312,-724,2866,-938,-1366,86,-585,1150,1637,627,500,364,713,726,1042,-231,616,243,1337,315,722,212,833,509,1128,741,763,-23,706,-315,392,-241,100,-771,-188,-212,298,-422,-413,-205,37,-281,-130,-523,-88,-503,-112,-573,-211,-454,-80,-504,-132,-294,-80,-239,-22,-204,-163,-320,-82,-296,-297,-476,-149,-313,-195,-414,-189,-328,-173,-333,-197,-351,-236,-353,-212,-335,-289,-358,-212,-232,-235,-269,-127,-247,-255,-385,-167,-387,-267,-392,-201,-373,-245,-334,-180,-331,-245,-322,-210,-238,-125,-198,-59,-138,-44,-137,-3,-58,9,-199,-115,-324,-112,-304,-122,-334,-195,-342,-88,-186,-146,-296,-157,-238,-181,-256,-164,-321,-198,-320,-193,-346,-189,-318,-169,-297,-240,-369,-178,-273,-261,-370,-222,-299,-226,-327,-186,-311,-192,-283,-157,-277,-144,-267,-140,-249,-109,-220,-95,-203,-109,-172,-51,-152,-88,-149,-49,-133,-83,-123,-30,-107,-62,-145,-55,-94,-23,-81,-49,-86,-1,-42,-24,-119,-38,-58,-5,-105,-50,-71,1,-68,-36,-53,-6,-19,20,-14,18,-29,55,-17,56,-9,73,3,81,35,97,32,73},
+   /* IRC_Composite_C_R0195_T045_P030.wav */
+   {9,-7,5,-3,0,3,-7,12,-18,25,-34,45,-60,79,-107,151,-363,-677,-240,-871,518,-941,-269,-1276,265,-1859,1187,-1350,2149,1,5698,-4523,4866,-12427,-2128,5167,3666,25146,2435,401,417,1825,-3164,2605,613,-1851,-1719,979,-329,2301,-426,1606,-126,1675,119,1149,-57,595,-80,1644,19,793,-129,1326,678,1045,-17,856,-22,267,-380,190,-844,-146,-664,199,-735,-167,-428,32,-402,-217,-411,-308,-427,-284,-543,-438,-454,-307,-561,-259,-430,-199,-381,-181,-382,-148,-228,-87,-405,-169,-303,-91,-361,-175,-378,-291,-318,-127,-301,-269,-260,-171,-316,-164,-268,-241,-361,-178,-341,-229,-322,-201,-335,-221,-314,-343,-328,-221,-321,-372,-308,-210,-400,-380,-322,-269,-386,-304,-356,-223,-292,-166,-317,-192,-91,-100,-158,-51,-73,-116,-23,-56,-150,-243,-92,-175,-169,-181,-177,-294,-169,-187,-210,-286,-130,-227,-213,-253,-159,-238,-150,-211,-249,-260,-144,-232,-247,-283,-227,-289,-178,-293,-252,-299,-148,-285,-211,-271,-197,-276,-200,-222,-204,-252,-176,-235,-205,-223,-107,-218,-119,-198,-109,-182,-35,-149,-101,-171,-62,-123,-44,-128,-75,-141,-37,-172,-54,-143,-11,-144,-27,-104,26,-103,23,-105,37,-73,18,-74,15,-63,4,-83,16,-68,-5,-50,47,-28,24,-35,41,-27,78,-21,77,-17,105,16,96,7,122,24},
+   /* IRC_Composite_C_R0195_T060_P030.wav */
+   {8,-7,7,-6,5,-5,4,-3,3,-3,3,-3,5,-9,20,86,-790,103,-872,55,-678,-196,-962,-387,-893,162,722,1609,608,2929,1713,-6837,-667,-12691,15713,13906,9242,5982,83,-1809,1076,1596,1326,-1529,-1392,-366,731,78,1368,-246,850,592,521,699,1115,310,930,973,592,664,159,708,710,489,-74,485,-697,374,-695,249,-1252,204,-915,137,-777,-91,-241,-403,-350,-466,-269,-521,-499,-540,-380,-320,-709,-233,-517,-228,-634,-244,-652,-290,-447,-147,-545,-241,-352,5,-488,-69,-437,53,-465,-6,-353,-78,-525,-24,-335,-157,-418,-27,-367,-103,-287,-241,-397,-158,-276,-304,-351,-181,-369,-225,-401,-209,-512,-184,-447,-153,-476,-152,-484,-76,-492,-243,-498,-127,-515,-186,-392,-135,-408,-53,-239,-82,-191,13,-138,-32,-118,-6,-177,-44,-187,-143,-246,-50,-262,-108,-264,-71,-256,-48,-325,-90,-349,-63,-352,-110,-280,-159,-311,-153,-350,-187,-295,-152,-318,-137,-192,-178,-252,-142,-284,-157,-277,-190,-286,-151,-282,-119,-257,-91,-310,-54,-224,-93,-207,-75,-245,-9,-203,-48,-237,12,-182,-40,-201,33,-146,-26,-200,-37,-156,-43,-159,-28,-130,-32,-114,20,-143,6,-113,12,-126,58,-103,21,-121,59,-91,44,-108,59,-43,47,-58,60,-38,52,-61,83,-36,62,3,66,27,84,84,65,66,103},
+   /* IRC_Composite_C_R0195_T075_P030.wav */
+   {100,-103,106,-109,112,-115,118,-120,122,-122,119,-110,88,-25,53,-874,378,-897,369,-1226,537,-1263,224,-1798,1274,98,2195,-462,4170,361,-3989,-3149,-11328,17722,9812,11683,3944,-511,-808,1931,2611,1751,-2704,-605,-430,728,-349,908,612,941,-687,931,759,-66,894,2123,310,1489,449,746,475,494,-493,-725,213,-1074,381,-1100,-205,-670,-115,-645,-170,-469,-431,31,-824,-248,-707,-458,-476,-267,-901,-228,-228,-601,-420,-335,-514,-545,-367,-632,-446,-566,-351,-391,-306,-444,-134,-388,-144,-333,-243,-194,-98,-434,-105,-173,-206,-249,-121,-377,-102,-345,-159,-256,-146,-470,-106,-327,-152,-330,-205,-368,-192,-409,-197,-444,-310,-359,-229,-461,-161,-393,-318,-351,-227,-387,-261,-314,-291,-270,-249,-242,-213,-299,-192,-269,-108,-253,-73,-278,80,-171,-11,-111,7,-198,-30,-155,-148,-117,-164,-168,-181,-155,-172,-221,-156,-207,-176,-229,-130,-292,-171,-280,-233,-288,-132,-237,-242,-209,-166,-245,-147,-263,-160,-240,-126,-261,-109,-276,-150,-274,-92,-247,-135,-225,-43,-207,-57,-174,-33,-157,-10,-172,-20,-116,-56,-146,22,-132,-61,-119,-53,-166,-22,-107,-81,-99,-40,-65,-89,-70,-32,-77,-31,-75,-25,-75,36,-146,3,-52,3,-100,13,-55,-5,-17,13,-41,5,-12,59,-13,32,3,61,44,107,53,129,85,118,95},
+   /* IRC_Composite_C_R0195_T090_P030.wav */
+   {-85,89,-94,98,-102,107,-112,116,-119,121,-119,106,-58,-455,60,99,-1054,-36,-88,242,-1635,433,-1255,1196,-1210,2540,-840,5344,-2798,4670,-10472,-874,1281,9198,19443,266,2404,-1705,3816,1662,3038,83,-1499,-2490,1368,-37,500,-334,1490,-626,909,-15,1075,678,1886,658,852,386,1418,146,400,-1073,-338,-633,-165,-1024,-383,-1251,-264,-407,-25,-1062,-140,-559,70,-716,-459,-642,-96,-725,-472,-412,-384,-647,-281,-605,-395,-572,-399,-706,-351,-583,-297,-495,-434,-423,-201,-406,-407,-323,-268,-341,-186,-360,-201,-241,-141,-313,-94,-257,-169,-208,-194,-224,-147,-275,-276,-255,-215,-224,-195,-256,-265,-271,-252,-352,-290,-430,-350,-309,-268,-387,-267,-289,-271,-262,-303,-332,-237,-287,-328,-292,-220,-307,-211,-297,-184,-216,-183,-248,-116,-180,-163,-187,-112,-160,-80,-117,-46,-81,-25,-91,-53,-114,-120,-176,-181,-173,-184,-205,-224,-154,-233,-229,-178,-204,-243,-197,-188,-194,-206,-154,-204,-205,-213,-116,-191,-127,-210,-145,-216,-173,-183,-177,-263,-209,-141,-116,-148,-107,-124,-68,-87,-45,-121,-68,-134,-77,-87,-29,-115,-67,-117,-46,-114,-27,-115,-45,-100,-16,-82,-32,-85,-17,-74,-28,-41,-27,-85,-23,-68,-40,-47,-15,-96,-13,-38,11,-39,-38,-19,1,-19,-3,-8,-15,-22,28,17,70,76,128,80,124,112,151},
+   /* IRC_Composite_C_R0195_T105_P030.wav */
+   {104,-111,117,-125,133,-143,153,-166,180,-197,218,-245,288,-401,132,-429,-287,-538,503,-701,-185,-1083,867,-1151,1236,-192,2589,704,2171,-548,-4634,-1245,-3979,18483,6281,7009,-196,501,1762,4063,3838,-1,-2732,-1085,-409,884,444,-393,-9,733,362,138,1179,1594,1300,947,317,734,700,55,-900,-943,-310,-712,-521,-933,-377,-1554,-143,-410,48,-924,-152,-440,-212,-554,-286,-601,-527,-371,-454,-653,-533,-426,-450,-579,-486,-483,-369,-609,-354,-543,-201,-457,-293,-586,-146,-430,-339,-522,-196,-439,-324,-340,-141,-423,-157,-344,-39,-390,-88,-363,-66,-244,-152,-284,-176,-153,-268,-264,-245,-148,-256,-290,-221,-325,-190,-340,-211,-490,-165,-333,-238,-471,-203,-235,-228,-397,-218,-264,-147,-365,-267,-367,-142,-319,-211,-356,-158,-290,-141,-243,-127,-217,-86,-135,-64,-128,-75,-153,-67,-155,-25,-190,-45,-193,-18,-259,1,-235,-77,-284,-88,-227,-151,-250,-186,-245,-111,-256,-126,-274,-46,-272,-88,-254,-32,-214,-71,-201,-144,-187,-98,-154,-182,-196,-78,-154,-85,-159,-36,-221,-26,-121,-4,-188,-47,-127,-22,-132,-24,-158,-49,-155,-2,-158,-21,-135,-30,-121,10,-87,-16,-118,10,-48,-22,-86,-12,-80,14,-80,13,-86,50,-103,29,-96,64,-85,45,-104,50,-45,64,-67,-11,-39,117,-14,158,27,234,13,225,68},
+   /* IRC_Composite_C_R0195_T120_P030.wav */
+   {28,-28,29,-29,29,-29,29,-27,25,-21,15,-5,-12,43,-140,88,-559,-189,139,-567,198,-314,-739,-383,1538,-450,794,642,4608,-2302,1277,-7331,2563,5148,5379,12280,2713,-323,-1693,7682,3305,1850,-1530,-2098,-1411,1431,-392,275,-715,741,-246,1255,618,1870,987,1146,368,730,-169,18,-891,-805,-1522,-439,-421,-616,-1006,-319,-441,-263,-511,-164,-630,-92,-636,-184,-746,-202,-725,-323,-658,-347,-529,-471,-673,-383,-415,-310,-627,-358,-521,-225,-567,-258,-424,-264,-432,-273,-312,-333,-413,-308,-330,-340,-367,-280,-273,-329,-276,-223,-220,-254,-266,-200,-177,-199,-276,-232,-217,-244,-204,-238,-182,-299,-214,-297,-195,-311,-235,-299,-177,-299,-241,-285,-222,-309,-277,-284,-244,-293,-235,-236,-241,-258,-206,-250,-223,-292,-255,-245,-224,-254,-179,-205,-152,-237,-116,-146,-78,-191,-75,-140,-27,-176,-63,-134,-82,-199,-79,-164,-84,-202,-123,-164,-105,-201,-110,-207,-105,-211,-126,-202,-134,-203,-129,-204,-131,-154,-100,-194,-94,-163,-74,-166,-82,-216,-107,-149,-27,-161,-56,-129,-40,-130,-55,-114,-53,-102,-63,-77,-31,-89,-106,-127,-83,-123,-74,-94,-24,-173,-31,-94,11,-133,-8,-83,-23,-88,12,-65,-46,-111,25,-94,13,-117,28,-88,37,-79,24,-35,51,-27,22,7,50,-3,53,7,103,86,163,91,162,75},
+   /* IRC_Composite_C_R0195_T135_P030.wav */
+   {6,-6,7,-7,8,-8,9,-9,10,-10,11,-12,13,-14,16,-18,-114,-234,-98,-110,-115,-173,-170,-232,-274,363,317,1353,133,2229,1323,703,-7423,4287,1521,5913,8831,4880,804,-872,5713,3941,3130,-2817,-2290,-315,304,-50,-26,-209,494,295,1100,751,1288,533,716,964,54,-449,-399,-528,-808,-934,-947,-951,-457,-538,-157,-625,-308,-385,-95,-571,-310,-374,-278,-494,-514,-429,-461,-556,-433,-529,-432,-535,-327,-580,-236,-492,-287,-555,-216,-430,-280,-451,-246,-351,-270,-341,-290,-325,-254,-404,-265,-383,-232,-420,-232,-398,-198,-399,-199,-300,-210,-291,-213,-230,-263,-273,-261,-215,-220,-262,-203,-257,-189,-275,-199,-298,-243,-292,-198,-258,-246,-263,-220,-238,-243,-271,-259,-242,-227,-263,-220,-247,-166,-281,-180,-274,-165,-251,-184,-245,-176,-216,-217,-207,-158,-156,-155,-166,-106,-127,-108,-139,-81,-152,-98,-167,-88,-170,-109,-150,-125,-150,-119,-119,-124,-135,-111,-142,-113,-144,-112,-168,-76,-193,-154,-199,-103,-124,-149,-137,-125,-90,-105,-96,-120,-90,-84,-94,-99,-96,-69,-90,-69,-82,-94,-80,-67,-60,-94,-70,-110,-76,-102,-51,-103,-55,-74,-38,-105,-57,-91,-76,-96,-51,-36,-59,-52,-57,-56,-61,-46,-70,-31,-32,-25,-47,6,-6,17,-33,-1,20,49,32,43,24,63,112,138,114,43,112},
+   /* IRC_Composite_C_R0195_T150_P030.wav */
+   {40,-42,44,-47,49,-53,56,-60,64,-69,75,-82,90,-99,110,-125,148,-207,85,-316,199,-301,148,-341,137,-377,122,76,967,463,1542,593,2935,-2335,-3746,3892,1126,7018,5853,6116,-1287,2364,4148,3574,181,-2393,-1125,-178,512,-398,-17,-462,310,1083,929,1228,538,479,431,450,-124,-446,-926,-681,-484,-854,-858,-886,-425,-380,-163,-187,-475,-333,-267,-193,-398,-286,-470,-331,-472,-511,-390,-535,-398,-443,-447,-458,-316,-355,-323,-348,-312,-366,-331,-282,-304,-410,-276,-324,-270,-334,-288,-347,-263,-345,-283,-348,-294,-324,-322,-384,-264,-346,-287,-352,-258,-326,-246,-345,-247,-332,-220,-276,-243,-276,-171,-257,-208,-277,-184,-237,-184,-265,-186,-245,-165,-259,-201,-253,-180,-238,-198,-272,-176,-246,-187,-277,-161,-252,-150,-275,-137,-267,-126,-254,-141,-241,-149,-227,-151,-206,-151,-203,-126,-181,-101,-170,-85,-185,-87,-146,-87,-170,-106,-170,-90,-160,-111,-150,-107,-140,-90,-144,-63,-132,-90,-238,-87,-160,-61,-203,-110,-165,-57,-102,-36,-116,-53,-98,-41,-96,-62,-105,-61,-102,-70,-120,-86,-109,-103,-103,-70,-67,-72,-85,-74,-124,-64,-94,-56,-139,-102,-99,-75,-97,-58,-61,-30,-94,-25,-74,-43,-104,-48,-76,-56,-75,-27,-39,-24,-3,21,-10,-1,6,97,22,44,1,101,93,134,63,82,45},
+   /* IRC_Composite_C_R0195_T165_P030.wav */
+   {-5,5,-6,6,-6,6,-6,7,-7,7,-8,8,-9,9,-10,11,-12,14,-17,25,-163,17,-80,12,2,-73,14,-200,92,163,755,471,1371,723,2348,-678,-3403,2150,1314,5172,5430,6608,508,1066,3048,2691,1177,-1954,-1312,-864,1114,-99,-395,-170,755,800,840,925,689,-7,-2,-92,-86,-565,-703,-538,-415,-580,-306,-607,-337,-594,-235,-197,-71,-393,-276,-461,-131,-367,-319,-372,-331,-590,-309,-448,-316,-391,-370,-330,-202,-345,-331,-285,-252,-333,-332,-343,-285,-341,-299,-355,-341,-319,-335,-324,-306,-321,-304,-329,-296,-343,-321,-372,-317,-381,-294,-371,-299,-341,-289,-334,-273,-311,-239,-276,-234,-285,-217,-256,-202,-250,-169,-221,-150,-205,-167,-217,-204,-247,-225,-236,-214,-247,-214,-207,-155,-197,-183,-192,-155,-160,-191,-236,-192,-219,-188,-240,-195,-229,-141,-216,-140,-210,-108,-198,-110,-186,-131,-164,-118,-161,-128,-132,-117,-113,-132,-130,-137,-165,-120,-157,-110,-125,-107,-149,-159,-131,-114,-123,-178,-132,-136,-75,-82,-32,-23,-71,-33,-49,-17,-94,-64,-87,-53,-94,-80,-79,-100,-87,-113,-66,-94,-109,-102,-113,-90,-94,-92,-105,-128,-124,-129,-89,-94,-67,-93,-83,-95,-80,-94,-87,-89,-91,-53,-81,-31,-57,18,-49,-13,-37,25,-7,9,-13,-1,53,58,59,79,109,133,124,84,60},
+   /* IRC_Composite_C_R0195_T180_P030.wav */
+   {11,-12,12,-12,13,-13,14,-14,15,-15,16,-17,18,-19,20,-21,23,-25,28,-33,39,-51,85,16,-87,129,-81,94,33,-96,-40,382,84,940,857,674,1733,496,-329,-2997,2450,4178,3824,5230,3832,1334,27,2852,1755,841,-2811,-750,709,798,-492,-489,295,1105,873,238,383,361,52,-483,-293,-277,-430,-468,-128,-283,-308,-272,-363,-267,-297,-189,-289,-143,-258,-242,-373,-291,-355,-331,-315,-246,-385,-325,-276,-197,-314,-270,-313,-267,-313,-307,-364,-269,-366,-341,-361,-328,-371,-363,-339,-357,-363,-329,-335,-334,-353,-302,-358,-291,-419,-330,-381,-332,-384,-319,-331,-309,-321,-276,-296,-256,-286,-256,-306,-268,-277,-214,-200,-211,-199,-179,-152,-213,-196,-243,-196,-273,-196,-241,-183,-226,-151,-195,-173,-198,-169,-171,-172,-166,-179,-179,-198,-192,-224,-204,-227,-203,-207,-198,-181,-177,-142,-148,-122,-136,-110,-124,-113,-120,-142,-140,-137,-157,-161,-213,-116,-179,-81,-164,-112,-182,-121,-113,-114,-132,-189,-130,-138,-73,-121,-47,-93,-7,-49,37,-29,-7,-77,-47,-80,-70,-85,-99,-107,-91,-49,-61,-81,-124,-69,-122,-37,-160,-57,-192,-64,-193,-77,-181,-87,-158,-111,-159,-105,-113,-89,-109,-88,-83,-83,-88,-75,-44,-73,-39,-53,-1,-6,52,20,54,20,68,72,99,107,107,128,102,141,66},
+   /* IRC_Composite_C_R0195_T195_P030.wav */
+   {-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,1,-1,1,-1,0,0,-1,2,-3,4,-1,20,-103,88,-97,77,-98,0,-92,123,108,628,541,868,946,950,-323,-2426,1960,1868,3655,4432,4145,1520,425,2257,2243,324,-1505,-101,360,566,-340,-97,521,852,598,427,177,417,31,149,-102,-24,-395,-11,-98,-149,-157,-72,-31,-41,-10,-27,-82,-183,-287,-120,-245,-255,-309,-131,-235,-200,-246,-222,-253,-210,-246,-249,-280,-281,-300,-278,-328,-316,-325,-352,-332,-302,-334,-406,-331,-345,-314,-353,-298,-355,-316,-331,-341,-377,-390,-356,-375,-325,-347,-301,-303,-263,-297,-274,-288,-262,-288,-280,-284,-231,-214,-205,-233,-199,-221,-205,-231,-218,-253,-209,-218,-203,-211,-199,-179,-191,-186,-199,-201,-193,-181,-187,-195,-182,-193,-188,-207,-197,-194,-197,-186,-192,-163,-178,-160,-165,-142,-165,-124,-155,-125,-159,-112,-160,-116,-190,-111,-165,-107,-172,-122,-190,-155,-169,-127,-164,-144,-157,-122,-135,-85,-113,-68,-116,-46,-67,-29,-54,-47,-65,-70,-92,-108,-92,-118,-99,-92,-87,-77,-99,-78,-106,-86,-118,-92,-123,-105,-136,-119,-126,-128,-143,-162,-171,-163,-154,-125,-147,-120,-134,-91,-125,-91,-103,-88,-80,-77,-45,-46,-3,16,24,24,27,58,42,84,85,129,127,141,127,125},
+   /* IRC_Composite_C_R0195_T210_P030.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,4,-4,4,-4,4,-4,5,-5,6,-6,7,-9,12,-23,-83,-41,-51,-44,-54,-36,-72,-62,13,62,362,515,745,618,964,123,-1474,242,1605,2935,3779,3927,2051,644,1570,1770,938,-679,-243,210,595,104,204,284,657,698,453,338,507,437,72,227,224,63,-55,88,141,215,131,225,97,158,60,38,-69,-107,-265,-85,-124,-202,-245,-131,-257,-191,-192,-178,-241,-145,-269,-233,-270,-235,-338,-254,-338,-292,-359,-271,-388,-290,-377,-344,-357,-293,-419,-370,-380,-349,-397,-346,-388,-357,-344,-302,-335,-306,-279,-276,-246,-273,-284,-282,-237,-264,-250,-226,-225,-248,-218,-230,-247,-240,-246,-259,-217,-216,-201,-233,-193,-207,-183,-205,-202,-228,-190,-207,-213,-222,-215,-237,-186,-220,-176,-235,-136,-217,-116,-201,-133,-200,-133,-159,-165,-150,-164,-152,-134,-165,-144,-175,-119,-164,-116,-139,-114,-152,-126,-132,-128,-147,-152,-173,-154,-145,-152,-115,-135,-100,-99,-65,-75,-79,-71,-101,-104,-126,-96,-135,-132,-139,-125,-80,-131,-108,-168,-90,-142,-81,-142,-146,-133,-121,-109,-142,-113,-121,-117,-148,-144,-150,-144,-150,-171,-140,-133,-106,-126,-118,-107,-114,-61,-79,-59,-59,-5,-31,13,-2,37,21,52,67,76,116,109,131,125},
+   /* IRC_Composite_C_R0195_T225_P030.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,2,-2,3,-3,3,-4,5,-7,14,-133,-89,-52,-68,-90,-53,-98,-77,-65,-141,-33,-38,144,467,407,586,730,265,-702,-561,1147,2053,3022,3625,2128,1090,1205,1750,1123,130,-65,370,545,406,83,446,636,787,529,661,549,526,605,368,384,101,455,156,304,149,290,238,156,207,48,113,-86,-57,-36,-176,-93,-129,-107,-209,-107,-181,-195,-158,-169,-157,-239,-205,-240,-206,-253,-259,-282,-324,-296,-377,-347,-379,-340,-400,-365,-377,-398,-363,-379,-335,-398,-330,-337,-283,-328,-250,-335,-246,-279,-228,-301,-284,-299,-253,-272,-251,-277,-231,-269,-213,-274,-207,-256,-219,-248,-212,-205,-234,-191,-244,-197,-238,-194,-226,-213,-226,-214,-216,-219,-201,-233,-193,-216,-184,-183,-195,-171,-198,-136,-183,-121,-177,-126,-151,-131,-156,-138,-160,-140,-152,-130,-143,-109,-146,-113,-168,-114,-165,-123,-191,-139,-186,-143,-163,-101,-126,-91,-147,-80,-140,-85,-158,-107,-181,-100,-149,-100,-152,-116,-152,-120,-144,-111,-159,-127,-155,-129,-148,-146,-173,-157,-145,-134,-136,-128,-146,-120,-128,-115,-125,-133,-149,-144,-153,-151,-147,-139,-128,-113,-110,-89,-78,-59,-67,-27,-39,-8,-19,9,9,34,44,63,65,98,113},
+   /* IRC_Composite_C_R0195_T240_P030.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,1,-97,-166,-68,-82,-98,-145,6,-178,-148,-79,-61,-4,226,289,466,695,150,-363,-564,127,1778,2371,2730,2418,1314,1333,1958,1573,362,208,618,691,672,358,520,767,841,691,635,738,675,472,476,447,387,278,358,303,322,169,236,146,167,153,93,161,-50,82,-137,22,-215,-48,-220,-102,-183,-156,-150,-189,-134,-246,-183,-293,-242,-320,-276,-333,-344,-324,-341,-336,-359,-336,-345,-340,-353,-365,-331,-325,-327,-353,-302,-293,-288,-330,-303,-271,-296,-272,-313,-253,-304,-231,-298,-228,-279,-211,-267,-230,-276,-219,-263,-229,-267,-208,-254,-187,-265,-150,-261,-152,-276,-148,-255,-165,-256,-184,-237,-198,-227,-197,-220,-180,-224,-157,-210,-127,-200,-118,-181,-83,-171,-100,-162,-99,-139,-124,-137,-122,-137,-107,-131,-113,-152,-146,-173,-133,-190,-141,-182,-105,-142,-132,-151,-164,-146,-160,-134,-193,-153,-203,-128,-188,-144,-178,-164,-173,-166,-129,-130,-161,-160,-179,-136,-174,-125,-201,-162,-176,-100,-152,-132,-174,-152,-170,-137,-167,-127,-157,-111,-151,-85,-120,-73,-114,-88,-97,-72,-75,-82,-80,-55,-47,-53,-51,-32,-31,18,-8,35,22,73,65},
+   /* IRC_Composite_C_R0195_T255_P030.wav */
+   {1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,2,-2,2,-2,3,-3,4,-4,5,-6,8,-12,21,-67,-186,-86,-78,-102,-151,-102,-59,-120,-188,-201,-77,-93,-133,190,315,388,642,206,-429,-627,-41,1512,2532,2996,2313,1394,1764,1723,1247,1241,976,437,321,813,834,517,405,589,1007,744,553,490,641,609,429,381,224,460,347,308,191,277,332,175,143,27,143,-12,2,-104,-86,-53,-114,-79,-254,-137,-211,-109,-349,-252,-297,-195,-348,-341,-330,-262,-295,-325,-316,-289,-220,-305,-292,-333,-256,-294,-316,-394,-312,-314,-286,-387,-317,-323,-233,-326,-302,-311,-250,-238,-292,-290,-296,-199,-259,-241,-292,-230,-245,-240,-270,-264,-223,-199,-230,-230,-203,-176,-179,-217,-226,-180,-203,-190,-239,-188,-230,-173,-226,-190,-237,-149,-184,-187,-190,-158,-126,-139,-130,-127,-103,-111,-83,-113,-124,-148,-136,-140,-136,-138,-118,-132,-127,-165,-140,-179,-161,-224,-175,-209,-159,-182,-149,-200,-170,-196,-150,-218,-152,-222,-177,-254,-168,-210,-186,-215,-137,-166,-128,-166,-128,-184,-106,-195,-174,-233,-137,-162,-153,-164,-155,-113,-146,-99,-141,-92,-113,-93,-86,-120,-80,-115,-69,-118,-75,-88,-73,-78,-77,-48,-78,-41,-54,-26,-50,-11,-21,13,14,31},
+   /* IRC_Composite_C_R0195_T270_P030.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,2,-2,4,-6,13,-118,-205,-234,-125,-147,-202,-210,-154,-149,-302,-174,-138,74,139,366,494,501,-30,-788,-660,656,2336,3548,2933,1466,1698,2087,1596,1331,660,745,348,609,649,510,592,471,837,634,881,692,658,657,499,658,390,588,291,406,283,323,211,84,205,87,193,12,79,-62,28,-174,-101,-266,-102,-288,-156,-317,-186,-313,-245,-324,-271,-272,-274,-260,-294,-238,-276,-248,-278,-258,-319,-271,-317,-281,-337,-312,-324,-337,-313,-331,-296,-324,-287,-323,-272,-287,-250,-300,-247,-285,-236,-292,-228,-244,-210,-272,-214,-253,-227,-239,-213,-239,-229,-229,-214,-226,-214,-215,-199,-227,-177,-203,-180,-193,-178,-164,-152,-146,-149,-125,-151,-141,-143,-150,-155,-139,-130,-168,-120,-114,-108,-126,-125,-154,-199,-182,-165,-181,-236,-178,-180,-150,-191,-167,-205,-200,-201,-200,-203,-212,-213,-204,-217,-180,-215,-180,-171,-178,-161,-169,-169,-194,-215,-198,-197,-181,-198,-150,-180,-147,-175,-139,-160,-138,-133,-143,-113,-144,-115,-133,-126,-110,-112,-106,-129,-97,-110,-96,-107,-98,-88,-83,-69,-93,-79,-84,-51,-65,-56,-47,-41,-18,-38,4,-20,19,6},
+   /* IRC_Composite_C_R0195_T285_P030.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,-1,1,-1,1,-1,1,-1,2,-2,3,-3,4,-7,13,-190,-183,-235,-156,-218,-208,-304,-188,-211,-199,-139,-161,153,336,233,-123,-164,-557,-685,1147,1908,2035,3160,1492,785,1242,1667,1506,818,1143,817,1178,780,1057,657,807,713,683,790,563,729,595,855,599,704,615,614,540,520,362,304,245,337,219,250,133,197,59,126,-65,-70,-160,-89,-159,-231,-254,-291,-203,-274,-185,-304,-208,-285,-189,-264,-256,-331,-243,-311,-262,-308,-280,-304,-246,-335,-250,-318,-269,-309,-282,-300,-260,-340,-268,-307,-236,-284,-269,-325,-251,-300,-247,-310,-244,-277,-183,-239,-198,-269,-229,-228,-237,-229,-245,-249,-264,-220,-228,-201,-208,-179,-162,-96,-70,-76,-134,-147,-121,-145,-142,-175,-192,-136,-108,-69,-153,-156,-210,-206,-236,-198,-320,-233,-263,-173,-277,-125,-224,-144,-166,-131,-224,-164,-197,-216,-260,-240,-244,-177,-196,-187,-157,-155,-171,-202,-195,-206,-203,-187,-195,-236,-200,-188,-186,-176,-232,-177,-163,-133,-189,-164,-179,-140,-153,-146,-152,-146,-135,-108,-114,-110,-119,-88,-135,-96,-118,-89,-127,-122,-110,-95,-72,-108,-98,-91,-59,-48,-71,-44,-52,-26,-30,-20,-36,-34,-3,-13,9,16},
+   /* IRC_Composite_C_R0195_T300_P030.wav */
+   {1,-1,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-3,3,-3,3,-3,3,-4,4,-4,5,-5,5,-6,7,-8,9,-10,12,-16,27,-117,-227,-314,-248,-219,-228,-232,-272,-389,-204,-111,300,56,366,86,-1011,-298,44,-215,2477,2714,2032,2711,645,499,536,1654,449,374,611,764,837,821,1350,980,1002,808,1149,967,761,841,707,633,745,777,758,579,642,512,570,444,429,371,457,296,276,181,191,36,17,10,-72,-108,-107,-232,-151,-205,-165,-300,-163,-265,-253,-269,-203,-258,-301,-275,-269,-292,-272,-260,-302,-297,-289,-229,-300,-264,-305,-243,-283,-264,-269,-279,-280,-287,-259,-269,-276,-247,-263,-262,-247,-231,-256,-218,-280,-241,-295,-239,-252,-218,-256,-230,-258,-241,-251,-148,-189,-159,-149,-45,-91,-37,-119,-83,-194,-47,-124,-58,-134,-130,-231,-259,-239,-276,-269,-285,-258,-248,-198,-188,-240,-206,-250,-202,-249,-205,-247,-163,-226,-240,-217,-219,-189,-209,-212,-237,-257,-142,-184,-180,-176,-167,-213,-202,-187,-188,-191,-235,-229,-241,-179,-197,-185,-210,-199,-189,-160,-163,-174,-186,-151,-189,-138,-133,-147,-130,-137,-107,-118,-101,-102,-120,-113,-110,-78,-102,-64,-110,-74,-76,-59,-92,-76,-85,-71,-79,-43,-88,-39,-85,-19,-81,13,-60,8,-26,31,8,16,23},
+   /* IRC_Composite_C_R0195_T315_P030.wav */
+   {-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-3,3,-4,4,-4,4,-4,5,-5,6,-7,10,-15,27,-80,-353,-110,-370,-400,-195,-345,-309,-394,-97,-188,193,754,56,-290,-194,-1922,214,1653,1159,4327,2754,1527,1212,743,317,539,1293,-56,-21,321,1262,295,442,708,1213,856,813,872,1000,917,1073,730,897,944,834,864,775,610,537,692,383,605,370,536,245,348,155,280,101,83,31,90,-127,-63,-179,-119,-272,-116,-304,-232,-233,-212,-345,-176,-295,-247,-334,-204,-342,-233,-195,-284,-324,-229,-236,-255,-304,-258,-323,-258,-297,-224,-323,-252,-263,-224,-317,-252,-224,-255,-281,-221,-218,-205,-288,-209,-253,-193,-244,-249,-201,-218,-210,-200,-172,-180,-181,-84,-203,-75,-54,19,-108,-2,-46,-171,-209,-203,-294,-294,-286,-264,-295,-154,-187,-222,-182,-252,-197,-249,-211,-305,-232,-289,-224,-284,-267,-256,-184,-221,-260,-295,-281,-237,-233,-228,-269,-251,-203,-162,-193,-192,-189,-166,-237,-188,-201,-208,-209,-181,-224,-228,-151,-192,-200,-209,-131,-196,-159,-162,-159,-166,-116,-160,-151,-131,-47,-142,-119,-122,-73,-93,-108,-96,-133,-90,-84,-86,-83,-90,-56,-96,-65,-65,-48,-80,-67,-79,-50,-60,-46,-63,-39,-44,-12,-27,9,-35,23,-3,24,6,41},
+   /* IRC_Composite_C_R0195_T330_P030.wav */
+   {-5,5,-5,5,-5,5,-5,5,-5,6,-6,6,-6,6,-6,6,-6,6,-7,7,-7,7,-8,8,-9,9,-11,14,-26,-447,-195,-107,-505,-457,-480,-118,-514,-260,-345,82,416,511,1319,-1763,-994,103,-1768,3014,4385,2231,5234,1274,290,-49,1679,111,-604,148,524,573,-122,973,360,1000,586,1035,552,578,788,885,854,618,749,894,854,1021,930,968,502,1006,765,768,438,460,339,388,236,299,159,177,-65,162,-6,-74,-134,-51,-230,-132,-193,-105,-327,-152,-257,-191,-291,-227,-315,-280,-252,-192,-306,-253,-281,-182,-326,-252,-283,-225,-353,-255,-282,-258,-322,-236,-271,-288,-246,-248,-299,-265,-254,-312,-330,-129,-295,-141,-204,-129,-190,-22,-154,-203,-125,-172,-156,-188,-111,-223,-112,-106,-241,-86,-37,-262,-238,-193,-103,-105,-7,-250,-241,-39,-63,-184,-238,-261,-364,-243,-263,-310,-361,-286,-407,-227,-301,-289,-311,-319,-263,-215,-201,-304,-287,-339,-226,-246,-239,-335,-276,-276,-181,-234,-181,-257,-180,-234,-151,-165,-203,-213,-194,-170,-205,-157,-164,-191,-168,-154,-86,-133,-128,-162,-96,-147,-81,-165,-68,-173,-90,-164,-64,-127,-68,-162,-97,-99,-50,-115,-84,-111,-102,-87,-74,-80,-123,-76,-57,-95,-59,-42,-45,-74,-38,-27,-9,-30,0,-55,36,-23,58,-21,61,-7,59,29},
+   /* IRC_Composite_C_R0195_T345_P030.wav */
+   {0,0,0,0,0,0,0,1,-1,1,-1,2,-2,2,-3,3,-4,5,-5,6,-7,8,-9,10,-10,-1,-563,-328,-175,-595,-424,-853,130,-981,36,-843,608,167,1104,1422,-2010,-2282,1073,-3818,6278,4848,2400,7303,1229,-443,-1130,2494,95,-616,-743,335,108,968,417,599,684,1073,630,810,221,839,469,1136,218,739,427,945,692,942,955,757,852,1008,762,614,598,745,200,403,51,357,29,333,-147,-65,9,35,-114,-161,-231,-52,-197,-139,-202,-61,-310,-211,-214,-165,-307,-207,-293,-261,-259,-204,-201,-310,-323,-244,-250,-260,-335,-226,-421,-206,-333,-191,-401,-188,-244,-264,-283,-225,-208,-338,-233,-69,-304,-181,-213,-18,-265,39,-68,-82,-113,-107,16,-329,-80,-324,-205,-399,-184,-328,-391,-87,-9,36,1,59,-85,124,-60,-143,-307,-231,-406,-296,-456,-311,-297,-269,-433,-350,-189,-283,-277,-303,-343,-325,-287,-213,-328,-275,-279,-293,-296,-268,-217,-284,-286,-294,-277,-236,-247,-201,-305,-195,-224,-135,-244,-185,-128,-225,-126,-186,-74,-235,-93,-209,-95,-133,-90,-131,-165,-80,-89,-69,-148,-104,-72,-157,-77,-134,-88,-171,-67,-113,-124,-110,-76,-110,-130,-59,-96,-88,-119,-61,-109,-81,-81,-67,-82,-54,-22,-49,-19,-18,2,-15,25,13,24,8,45,57,4,20,23,60}};
+
+const int16_t irc_composite_c_r0195_p045[][256] =
+  {/* IRC_Composite_C_R0195_T000_P045.wav */
+   {-30,30,-31,32,-33,34,-35,36,-37,38,-39,40,-41,42,-43,43,-44,44,-42,38,-28,-9,-568,-169,-659,-127,-629,-131,-879,-529,-702,-207,-686,-146,757,1500,46,1169,-4599,-1790,163,-4363,15423,4884,3940,951,722,-172,1158,1933,-227,-750,303,-613,1137,856,493,193,1077,347,1002,300,420,882,521,444,512,395,442,484,453,404,610,620,839,826,758,649,736,767,608,484,493,147,169,427,10,204,-311,19,-245,98,-241,0,-167,16,-127,-26,-105,-158,-191,-79,-246,-206,-268,-177,-349,-216,-328,-216,-366,-178,-332,-214,-308,-168,-352,-204,-312,-132,-301,-215,-309,-152,-304,-202,-336,-200,-308,-229,-340,-205,-331,-196,-326,-240,-274,-198,-310,-200,-321,-159,-282,-190,-264,-176,-207,-74,-246,-157,-167,-77,-245,18,-155,-133,-274,-40,-289,-273,-267,-139,-520,-214,-265,-252,-317,-46,-133,-17,38,121,-153,151,27,-13,-114,13,-297,-74,-188,-147,-332,-197,-268,-297,-199,-221,-310,-309,-263,-206,-258,-237,-414,-155,-291,-200,-367,-189,-287,-139,-241,-261,-247,-142,-190,-233,-239,-158,-249,-110,-242,-173,-207,-82,-217,-136,-180,-136,-164,-112,-177,-97,-134,-40,-164,-32,-115,-16,-91,-12,-112,4,-81,-19,-100,-20,-70,-21,-87,-6,-59,22,-68,-7,-66,36,-20,13,-47,-2,-27,28,-4,13},
+   /* IRC_Composite_C_R0195_T015_P045.wav */
+   {16,-14,12,-9,7,-4,1,3,-7,12,-17,22,-29,36,-45,56,-69,85,-107,143,-247,-403,-616,-230,-517,-270,-663,-670,-583,-556,-570,-300,850,1495,298,1490,-1579,-5393,1137,-9580,17573,8715,3526,4288,-490,-1083,280,3534,-555,-1021,-341,-114,804,1521,139,355,844,673,325,735,-35,1245,91,909,-18,912,280,465,82,792,554,703,257,849,594,952,381,794,169,438,96,219,18,183,-220,-11,-303,284,-345,-44,-429,-44,-355,-58,-224,-155,-342,-30,-264,-140,-360,-95,-355,-133,-396,-161,-354,-174,-312,-65,-314,-194,-357,-142,-289,-170,-390,-213,-337,-152,-369,-215,-364,-191,-332,-194,-330,-206,-298,-234,-318,-198,-344,-225,-368,-198,-333,-240,-308,-215,-230,-223,-291,-224,-258,-133,-220,-212,-260,-84,-207,-164,-294,-157,-248,-263,-285,-134,-218,-267,-295,-223,-237,-192,-163,-155,-68,86,-74,-67,-51,-46,-49,-45,-144,-181,-126,-51,-106,-136,-213,-189,-163,-189,-238,-230,-238,-276,-288,-300,-274,-264,-268,-228,-288,-202,-201,-180,-232,-217,-238,-174,-222,-220,-242,-187,-203,-190,-211,-143,-223,-148,-187,-110,-202,-179,-175,-124,-140,-207,-165,-148,-117,-138,-129,-91,-111,-52,-96,-31,-72,-57,-38,-41,-39,-63,-12,-65,-2,-67,12,-64,37,-48,34,-49,20,-4,26,-16,4,-9,34,-12,34,-23},
+   /* IRC_Composite_C_R0195_T030_P045.wav */
+   {53,-54,56,-58,59,-62,64,-67,70,-73,77,-81,87,-93,100,-110,122,-140,176,-592,-266,-620,-93,-807,111,-1033,-244,-1451,661,-1378,1038,309,3279,-1940,4376,-7187,-1452,-4658,38,22380,4913,3629,1106,-370,-1337,2958,848,-390,-2691,1295,-1155,2599,-36,956,635,787,53,821,200,581,67,983,274,565,247,646,211,751,653,273,546,759,707,776,230,704,98,258,-440,41,-44,-99,-570,-276,125,-89,-316,-287,-67,-256,-409,-175,-243,-300,-369,-371,-102,-291,-350,-382,-109,-199,-343,-360,-210,-168,-316,-283,-194,-249,-97,-252,-166,-374,-161,-245,-317,-297,-364,-202,-337,-267,-324,-305,-165,-347,-151,-302,-171,-371,-213,-207,-334,-383,-313,-191,-297,-398,-235,-273,-140,-391,-214,-292,-119,-316,-254,-221,-194,-227,-232,-269,-229,-362,-187,-364,-206,-335,-144,-247,-146,-237,-158,-174,-97,-117,50,-80,66,-91,178,-80,-68,-202,-96,-198,-132,-223,-177,-186,-153,-217,-140,-249,-182,-298,-91,-285,-230,-334,-153,-250,-218,-295,-154,-260,-140,-294,-141,-233,-87,-273,-133,-222,-68,-214,-169,-225,-107,-209,-114,-264,-92,-242,-49,-295,-99,-226,-66,-252,-85,-193,-65,-189,-57,-171,-14,-172,-4,-179,85,-157,3,-134,92,-113,-4,-109,69,-51,13,-79,68,-27,61,-58,66,-43,79,-23,54,-24,30,-27,66,-41},
+   /* IRC_Composite_C_R0195_T045_P045.wav */
+   {-71,74,-76,79,-82,84,-87,90,-93,96,-99,101,-102,102,-97,85,-52,-97,-607,-227,-812,438,-1024,-204,-760,95,-1636,140,-766,2279,-763,3056,150,2120,-5760,-6480,1711,-14,24602,5276,1036,-813,1031,1166,848,674,-478,-1487,-945,687,1855,-292,799,1002,786,207,559,505,363,364,343,530,502,11,500,785,837,217,463,907,599,564,357,380,-197,84,-367,169,-530,-810,-282,-180,-171,-519,-179,-442,-72,-160,-239,-494,-294,-88,-553,-268,-512,-278,-503,-198,-473,-229,-317,-213,-370,-187,-331,-249,-344,-154,-381,-135,-317,-147,-279,-176,-355,-133,-274,-305,-330,-235,-314,-312,-303,-290,-257,-159,-269,-234,-243,-173,-350,-204,-383,-239,-309,-233,-404,-183,-289,-321,-344,-212,-319,-251,-253,-168,-303,-204,-277,-189,-284,-216,-333,-204,-348,-208,-298,-166,-339,-189,-186,-273,-152,-87,-184,-166,10,48,-40,2,0,-18,-97,-3,-134,-124,-166,-216,-161,-149,-234,-162,-175,-239,-198,-150,-294,-241,-224,-190,-303,-189,-150,-236,-232,-162,-159,-199,-136,-231,-166,-169,-136,-155,-152,-125,-146,-133,-100,-123,-156,-121,-135,-145,-164,-174,-157,-151,-189,-118,-198,-112,-121,-108,-138,-101,-93,-63,-79,-82,-83,-26,-44,-58,-96,20,-91,-15,-43,22,-69,29,5,6,-19,24,3,1,25,26,26,17,31,13,22,22},
+   /* IRC_Composite_C_R0195_T060_P045.wav */
+   {59,-58,56,-55,52,-49,45,-40,34,-25,14,1,-23,55,-108,214,-619,-300,-123,-537,-172,-393,-415,-586,-104,-1306,69,-335,1375,535,2761,-686,3824,-8072,-1237,-7735,11382,19138,4865,2658,-2047,1916,571,2806,-624,-694,-1625,70,232,950,-64,1404,-414,1196,34,1403,-261,1089,343,834,-188,587,-251,1086,518,827,135,641,494,842,-31,161,-360,-291,-317,171,-783,-241,-1031,130,-617,-201,-704,-152,-418,-199,-454,-247,-285,-342,-410,-378,-429,-306,-467,-327,-419,-184,-471,-188,-474,-157,-446,-174,-469,-146,-371,-239,-326,-267,-268,-265,-279,-218,-230,-216,-236,-148,-289,-212,-358,-216,-330,-124,-259,-173,-298,-166,-287,-241,-319,-291,-323,-261,-320,-293,-286,-298,-318,-332,-305,-269,-323,-212,-266,-206,-301,-143,-304,-236,-268,-212,-344,-239,-265,-225,-364,-157,-292,-187,-206,-129,-241,-186,-152,-141,-106,-103,-15,-115,56,-68,26,-47,-92,-125,-141,-154,-176,-196,-181,-250,-197,-136,-112,-240,-89,-180,-145,-235,-146,-223,-248,-178,-229,-152,-219,-112,-250,-98,-141,-152,-155,-116,-137,-150,-110,-137,-128,-94,-113,-72,-189,-29,-176,-93,-125,-129,-123,-190,-65,-169,-92,-169,-68,-159,-48,-109,-46,-109,-25,-81,-31,-72,-27,-71,7,-65,-16,-30,-30,-44,30,-39,8,-56,53,-36,81,-51,97,-64,95,-35,60,-30},
+   /* IRC_Composite_C_R0195_T075_P045.wav */
+   {7,-7,6,-6,5,-5,4,-3,2,0,-2,5,-9,14,-23,45,-729,-86,-463,-18,-765,398,-1336,518,-1262,-130,-544,1878,-781,3861,-694,2694,-2479,-6040,-3168,1816,21907,5475,5383,-2274,1082,1645,3196,-180,-862,-1506,-288,-182,870,-91,1493,-388,779,238,870,64,614,1007,850,122,95,413,1397,214,480,297,844,-36,272,27,-433,-791,-467,60,-441,-446,-697,-441,-417,-402,-239,-587,-268,-547,-343,-464,-296,-435,-388,-347,-416,-430,-420,-403,-296,-406,-376,-408,-228,-422,-330,-353,-273,-416,-325,-234,-301,-372,-327,-219,-336,-296,-307,-215,-240,-202,-250,-161,-199,-207,-315,-154,-247,-142,-267,-158,-266,-221,-313,-201,-303,-302,-339,-264,-278,-318,-314,-301,-319,-312,-315,-204,-334,-223,-340,-127,-300,-165,-343,-198,-307,-246,-285,-207,-293,-254,-293,-183,-246,-202,-252,-100,-213,-78,-138,-60,-197,-164,-160,-109,-141,-50,-142,2,-155,50,-176,0,-217,-67,-265,-39,-241,-180,-198,-196,-186,-182,-111,-135,-134,-125,-176,-98,-184,-100,-212,-114,-228,-113,-95,-140,-143,-141,-78,-133,-100,-67,-178,-75,-161,-2,-319,35,-236,-7,-269,53,-223,-47,-143,-20,-108,-80,-80,-54,-96,-67,-96,-24,-164,-15,-108,-24,-125,-8,-47,-26,-57,4,-8,-39,-35,32,-69,30,-49,58,-82,89,-80,107,-80,104,-56,112},
+   /* IRC_Composite_C_R0195_T090_P045.wav */
+   {82,-84,86,-88,90,-91,93,-94,95,-95,94,-92,87,-78,61,-33,-240,-426,56,-683,113,-744,373,-1306,284,-1194,1644,-667,2652,-291,4245,-2423,-2014,-4070,-4168,13696,10251,10475,-3031,1940,668,3850,1809,-625,-1492,-994,-456,184,563,568,-96,1027,330,539,32,648,1068,571,102,392,437,1555,535,636,312,223,-470,382,-426,-676,-1277,-176,-342,-251,-706,-452,-477,-459,-100,-511,-346,-481,-454,-509,-470,-397,-448,-337,-587,-233,-611,-143,-563,-280,-521,-364,-455,-244,-466,-275,-326,-321,-388,-219,-441,-210,-415,-297,-415,-244,-390,-213,-373,-184,-325,-193,-152,-257,-243,-230,-170,-254,-69,-214,-166,-228,-242,-227,-305,-281,-292,-232,-343,-218,-305,-280,-254,-331,-243,-363,-249,-287,-245,-320,-255,-262,-272,-215,-275,-191,-269,-200,-292,-200,-287,-217,-267,-214,-217,-218,-189,-189,-148,-203,-120,-154,-118,-165,-102,-132,-161,-134,-144,-132,-168,-94,-134,-87,-100,-96,-67,-104,-58,-178,-114,-154,-134,-226,-145,-224,-215,-130,-174,-117,-174,-71,-88,-63,-109,-58,-78,-120,-50,-84,-79,-101,-63,-97,-110,-65,-129,-123,-160,-83,-162,-109,-152,-92,-143,-76,-108,-73,-110,-42,-74,-64,-99,-35,-69,-11,-79,-57,-78,-51,-41,-69,-60,-68,-23,-64,48,-114,27,-96,75,-109,79,-84,99,-121,109,-64,69,-52,115,-98},
+   /* IRC_Composite_C_R0195_T105_P045.wav */
+   {28,-30,31,-33,35,-38,40,-43,47,-51,55,-60,67,-75,85,-99,111,-72,-376,-88,-540,307,-524,-278,-931,669,115,481,924,2601,65,2397,-5614,-1324,-2647,10799,8614,10099,118,-2294,4895,2386,2569,-875,-924,-2723,797,264,147,-195,968,521,465,-376,1032,858,1253,-135,495,389,792,755,975,0,305,-239,-39,-629,-469,-1320,-587,-582,-54,-751,-376,-721,-115,-425,-260,-335,-443,-566,-430,-612,-353,-563,-240,-666,-197,-569,-261,-523,-199,-719,-213,-574,-181,-663,-112,-511,-96,-492,-191,-414,-208,-419,-314,-403,-269,-455,-207,-442,-146,-453,-78,-424,-85,-318,-146,-327,-204,-254,-94,-214,-119,-280,-105,-318,-172,-355,-253,-350,-232,-337,-213,-346,-174,-337,-196,-365,-187,-307,-214,-347,-231,-304,-233,-293,-202,-273,-150,-270,-125,-278,-116,-278,-165,-297,-205,-241,-198,-221,-169,-202,-83,-240,-100,-225,-103,-190,-146,-171,-133,-169,-147,-152,-62,-157,-83,-150,-63,-116,-82,-115,-86,-88,-91,-126,-129,-161,-111,-196,-127,-178,-79,-197,-117,-103,-73,-136,-26,-65,-11,-114,16,-69,41,-165,27,-83,-32,-83,-83,-89,-127,-138,-85,-115,-105,-151,-76,-90,-77,-130,-45,-81,-85,-105,-60,-35,-80,-62,-86,-27,-81,-18,-56,-46,-45,-2,5,-61,-31,-63,-14,-64,-36,8,-66,36,-43,73,-62,61,-61,80},
+   /* IRC_Composite_C_R0195_T120_P045.wav */
+   {-12,12,-13,13,-14,14,-15,16,-17,18,-20,22,-25,29,-34,41,-54,124,-315,231,-915,409,-381,76,-1011,336,-106,1161,-312,2462,834,1649,-538,-4939,-381,2968,10089,6155,8410,-2705,920,4167,3787,745,-2352,-1802,-28,292,-651,645,868,-434,709,424,388,1070,1285,-64,322,849,520,234,673,260,-252,-169,-219,-404,-898,-522,-1000,-760,-659,-37,-489,-625,-480,-117,-279,-318,-554,-444,-479,-476,-447,-397,-506,-426,-374,-367,-517,-341,-428,-392,-509,-321,-429,-316,-424,-291,-372,-230,-336,-266,-337,-314,-357,-245,-417,-326,-393,-223,-400,-276,-346,-207,-347,-196,-289,-214,-324,-195,-253,-149,-240,-179,-253,-165,-279,-187,-327,-224,-367,-235,-338,-247,-316,-185,-311,-218,-272,-162,-301,-187,-304,-196,-322,-148,-322,-157,-312,-104,-274,-142,-240,-141,-246,-193,-235,-171,-192,-186,-214,-189,-212,-181,-191,-123,-229,-156,-208,-159,-193,-117,-178,-146,-179,-127,-164,-97,-166,-64,-139,-51,-149,-26,-150,-13,-111,-43,-117,-85,-121,-102,-128,-113,-107,-143,-80,-75,-82,-86,-55,-52,-89,-61,-20,-13,-63,41,-66,-14,-65,-25,-99,-66,-90,-83,-119,-73,-60,-96,-94,-16,-118,-43,-110,-43,-94,-27,-153,-76,-126,-21,-94,-45,-92,-51,-33,3,-63,-11,-52,-9,-73,9,-41,-20,-40,3,-44,-34,0,0,-32,16},
+   /* IRC_Composite_C_R0195_T135_P045.wav */
+   {15,-16,17,-18,19,-20,21,-22,23,-24,25,-27,28,-29,29,-28,24,-11,-117,-145,-90,-121,-109,-173,-39,-435,83,-128,1107,344,1609,548,3339,-3005,-2346,-79,2825,6596,9304,4876,-1449,2789,1884,4671,84,-2193,-1608,-211,-70,580,333,233,35,298,-56,1401,612,725,299,829,803,425,-150,290,-261,-139,-438,-491,-740,-322,-692,-659,-1013,-400,-489,-130,-431,-461,-332,-161,-302,-454,-443,-489,-403,-443,-472,-442,-454,-357,-427,-305,-456,-382,-411,-333,-418,-365,-303,-376,-334,-316,-292,-311,-265,-339,-352,-281,-323,-329,-379,-310,-346,-316,-320,-334,-291,-274,-280,-297,-290,-272,-270,-230,-233,-229,-239,-263,-224,-254,-270,-293,-279,-264,-289,-263,-287,-260,-261,-232,-231,-207,-225,-198,-204,-183,-223,-189,-221,-183,-247,-188,-250,-189,-248,-190,-209,-183,-176,-163,-144,-170,-178,-173,-185,-158,-195,-164,-234,-222,-222,-206,-165,-198,-160,-175,-170,-167,-137,-130,-154,-126,-141,-104,-142,-91,-116,-80,-71,-71,-88,-42,-93,-59,-74,-48,-70,-66,-85,-79,-76,-90,-60,-94,-51,-65,-11,-46,-25,-16,-13,-19,-24,-51,-74,-67,-91,-21,-96,-77,-66,2,-73,-36,-56,-44,-134,-42,-35,-70,-118,-47,-68,-116,-72,-82,-75,-136,-24,-85,-36,-90,1,-62,-4,-20,-33,-14,1,-17,-33,-28,-35,-37,-9,-16},
+   /* IRC_Composite_C_R0195_T150_P045.wav */
+   {13,-14,16,-17,19,-20,22,-25,27,-30,34,-38,42,-48,55,-63,75,-91,119,-208,14,-179,43,-143,14,-237,127,-445,347,318,882,1092,671,2337,-2,-2406,-2033,3403,2686,9080,6065,1654,-385,2378,3537,1656,-964,-2514,-65,-61,604,313,-548,8,506,751,592,750,799,256,788,738,691,-486,-503,-169,-52,-560,-771,-513,-275,-418,-623,-527,-620,-550,-389,-73,-175,-357,-452,-335,-134,-523,-381,-574,-280,-554,-346,-501,-295,-395,-265,-376,-302,-406,-287,-380,-314,-400,-273,-369,-283,-407,-233,-368,-270,-408,-280,-353,-282,-379,-324,-362,-292,-349,-288,-317,-293,-299,-273,-329,-303,-342,-223,-328,-247,-359,-217,-332,-232,-362,-243,-293,-224,-266,-223,-255,-188,-248,-204,-263,-170,-234,-148,-228,-150,-221,-123,-200,-152,-223,-169,-222,-177,-244,-174,-210,-124,-206,-129,-206,-127,-177,-96,-151,-166,-237,-192,-200,-169,-204,-186,-233,-213,-217,-169,-178,-153,-156,-130,-156,-129,-136,-88,-131,-105,-128,-101,-129,-90,-83,-84,-106,-52,-43,-49,-65,-44,-66,-38,-70,26,-76,6,-88,-13,-82,-34,-41,-4,-34,-1,-14,9,-50,-23,-69,-16,-101,-10,-110,-17,-119,-37,-74,-19,-102,-24,-41,-56,-111,-39,-111,-124,-147,-53,-65,-109,-46,-41,-68,-52,-7,-29,-64,-43,-18,-36,-49,-47,1,-26,-14,-36,-17},
+   /* IRC_Composite_C_R0195_T165_P045.wav */
+   {-5,6,-6,7,-7,8,-9,9,-10,11,-13,14,-16,18,-20,23,-27,31,-37,45,-57,83,-131,98,-140,102,-154,88,-172,82,24,745,704,1159,942,1377,685,-3350,-75,2310,5071,6483,6504,-342,276,2310,3436,1307,-1968,-1879,-504,1149,580,-184,-486,250,728,683,585,651,265,901,267,694,-240,-105,-607,-269,-676,-500,-479,-378,-278,-303,-345,-367,-327,-471,-422,-217,-111,-233,-480,-413,-399,-332,-479,-354,-471,-387,-344,-260,-374,-267,-230,-298,-347,-353,-301,-344,-328,-357,-343,-372,-361,-346,-368,-366,-338,-358,-344,-365,-318,-353,-318,-336,-351,-319,-335,-289,-334,-309,-321,-326,-336,-348,-335,-354,-321,-327,-304,-329,-269,-270,-235,-245,-249,-208,-236,-185,-237,-214,-241,-196,-206,-169,-200,-166,-161,-150,-185,-187,-205,-193,-210,-166,-170,-138,-171,-155,-201,-170,-152,-109,-129,-189,-178,-185,-130,-167,-150,-206,-233,-222,-196,-166,-206,-190,-201,-159,-173,-138,-175,-147,-160,-114,-124,-105,-119,-77,-100,-117,-136,-90,-111,-91,-103,-87,-79,-69,-28,-22,-44,-41,-25,-6,-27,-13,-21,-10,-27,9,1,-13,14,-2,20,6,-17,-7,-23,-45,-82,-88,-89,-86,-91,-67,-101,-82,-96,-74,-71,-81,-64,-98,-60,-94,-66,-105,-103,-119,-65,-33,-28,-21,-24,-24,-66,-22,-25,-29,-63,-45,-61,-77},
+   /* IRC_Composite_C_R0195_T180_P045.wav */
+   {1,-1,1,-1,2,-2,2,-2,2,-2,2,-2,3,-3,3,-3,3,-4,4,-4,4,-4,3,-2,9,-9,70,-59,121,-126,127,-45,222,619,653,1289,587,1432,300,-1888,-1240,3381,3947,6141,5035,479,-148,2113,2998,773,-1353,-1664,690,1041,-58,-664,-159,711,746,445,284,522,787,289,291,216,96,-622,-296,-616,-684,-562,-153,-193,-156,-160,-230,-111,-119,-291,-584,-298,-144,-240,-398,-579,-379,-509,-208,-402,-311,-292,-227,-267,-259,-266,-213,-263,-327,-375,-294,-371,-362,-434,-364,-436,-381,-431,-375,-404,-373,-398,-329,-365,-326,-409,-345,-385,-333,-367,-288,-357,-297,-388,-309,-398,-314,-430,-330,-408,-271,-349,-243,-310,-230,-234,-197,-214,-201,-187,-203,-238,-232,-238,-212,-225,-205,-211,-176,-179,-201,-211,-228,-202,-224,-150,-202,-129,-214,-123,-184,-94,-133,-120,-193,-166,-163,-113,-144,-135,-197,-168,-225,-183,-213,-177,-199,-157,-207,-155,-195,-92,-173,-121,-200,-130,-155,-103,-148,-111,-166,-85,-143,-56,-147,-56,-154,-59,-141,-36,-120,-42,-95,-13,-67,16,-28,44,-44,24,-33,43,-27,82,-21,103,-17,122,-2,81,-69,27,-35,-19,-94,-112,-111,-109,-91,-94,-119,-134,-85,-80,-83,-129,-79,-97,-58,-112,-41,-119,-24,-104,-33,-99,-41,-79,-41,-68,-44,-36,-44,-71,-84,-70},
+   /* IRC_Composite_C_R0195_T195_P045.wav */
+   {-10,11,-11,11,-12,12,-12,13,-13,14,-14,15,-15,16,-17,17,-18,19,-20,21,-22,23,-24,26,-28,29,-70,1,21,-50,-12,7,-98,118,-221,432,295,945,530,962,1220,-497,-1827,293,2669,3717,6133,2421,667,647,2517,2286,876,-1019,-811,715,567,146,-552,68,761,748,406,192,794,584,428,293,114,-104,-113,-153,-376,-408,-234,-64,-66,-15,75,-43,-32,-84,-58,-289,-255,-381,-227,-301,-267,-362,-351,-257,-250,-265,-197,-240,-267,-205,-195,-288,-234,-313,-331,-391,-314,-376,-403,-414,-376,-388,-386,-415,-381,-386,-367,-392,-359,-382,-358,-395,-338,-337,-339,-329,-314,-348,-381,-383,-383,-385,-396,-353,-328,-277,-291,-244,-249,-220,-237,-207,-212,-209,-222,-227,-219,-186,-196,-201,-196,-190,-194,-210,-219,-243,-209,-201,-163,-195,-186,-186,-145,-123,-130,-154,-175,-167,-170,-122,-126,-124,-185,-179,-196,-160,-179,-161,-199,-183,-176,-134,-131,-147,-153,-166,-151,-161,-167,-166,-166,-160,-162,-139,-166,-135,-153,-143,-142,-125,-105,-102,-94,-90,-78,-72,-57,-50,-51,-21,-44,-9,-31,-16,-46,-3,-13,24,10,27,32,25,66,-13,-17,-41,-15,-51,-100,-128,-126,-123,-107,-116,-127,-132,-96,-99,-88,-127,-87,-121,-70,-130,-87,-96,-70,-95,-94,-64,-80,-73,-64,-54,-55,-52,-44,-57},
+   /* IRC_Composite_C_R0195_T210_P045.wav */
+   {-1,1,-1,1,-1,1,-1,1,-2,2,-2,2,-2,2,-2,3,-3,3,-3,4,-4,4,-5,5,-6,6,-7,9,-41,-44,-65,-30,-65,-8,-83,-60,-92,30,110,507,445,762,568,1174,-1167,-1019,628,2221,3797,4340,2039,369,1655,2128,2575,-74,-532,-44,828,327,-261,-60,394,940,522,463,465,701,615,239,186,177,226,24,55,-107,-121,-144,122,80,101,60,168,75,69,-42,-103,-208,-202,-275,-257,-213,-175,-274,-214,-201,-239,-203,-186,-264,-269,-249,-233,-294,-292,-329,-314,-341,-364,-402,-341,-398,-395,-448,-334,-429,-332,-431,-329,-423,-331,-395,-337,-340,-332,-347,-337,-368,-357,-389,-353,-376,-312,-331,-265,-307,-238,-251,-202,-250,-204,-228,-195,-243,-216,-235,-186,-234,-184,-216,-198,-218,-214,-243,-226,-238,-184,-210,-192,-215,-154,-146,-144,-168,-181,-177,-171,-140,-114,-169,-137,-193,-142,-178,-151,-186,-152,-183,-144,-144,-116,-143,-141,-149,-130,-174,-129,-185,-122,-193,-143,-180,-134,-169,-148,-166,-136,-141,-122,-130,-103,-114,-106,-102,-72,-91,-60,-67,-53,-70,-51,-40,-61,-20,-57,16,-56,22,-58,21,-49,-10,-48,3,-48,-36,-55,-46,-67,-83,-96,-101,-127,-82,-139,-122,-145,-101,-121,-109,-139,-130,-124,-118,-116,-120,-95,-97,-71,-73,-66,-58,-48,-24,-31,-11},
+   /* IRC_Composite_C_R0195_T225_P045.wav */
+   {-3,3,-3,3,-3,3,-3,4,-4,4,-4,4,-4,4,-4,5,-5,5,-5,5,-6,6,-6,7,-7,8,-9,10,-13,20,-75,-97,-66,-47,-85,-74,-101,-34,-123,-132,-22,222,292,490,466,903,275,-1049,-389,584,2358,3669,3152,1258,1261,1738,2331,1624,37,-28,186,674,40,219,313,646,713,656,528,603,598,473,348,444,144,373,151,223,11,95,112,186,158,153,282,115,221,30,18,-91,-32,-200,-106,-202,-164,-212,-141,-234,-274,-248,-205,-234,-252,-279,-220,-289,-259,-337,-286,-358,-337,-398,-361,-383,-377,-397,-403,-387,-416,-383,-391,-375,-383,-384,-351,-350,-337,-371,-325,-339,-314,-362,-307,-327,-273,-302,-260,-293,-265,-270,-253,-272,-249,-242,-222,-243,-203,-221,-193,-215,-188,-249,-219,-244,-222,-251,-227,-225,-219,-214,-206,-186,-149,-158,-170,-204,-178,-182,-146,-152,-156,-169,-175,-163,-157,-142,-159,-167,-156,-155,-165,-170,-148,-148,-131,-118,-114,-117,-133,-119,-145,-148,-151,-147,-145,-167,-157,-166,-139,-171,-158,-145,-135,-121,-133,-97,-119,-63,-84,-67,-101,-69,-67,-57,-75,-59,-62,-49,-63,-41,-77,-33,-68,-14,-83,-22,-64,-11,-77,-48,-83,-73,-129,-98,-117,-128,-152,-116,-144,-123,-154,-107,-170,-138,-138,-97,-115,-107,-82,-89,-65,-72,-83,-58,-72,-30},
+   /* IRC_Composite_C_R0195_T240_P045.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,2,-5,-83,-84,-72,-135,-73,-91,-109,-87,-133,-84,-173,-91,-36,199,307,354,685,469,-284,-594,-516,1131,3046,2851,2215,1571,1578,1884,1712,836,241,420,708,320,266,499,693,588,631,732,645,553,384,505,486,415,335,428,309,250,327,166,248,139,364,147,249,184,201,90,89,-4,-84,-57,-140,-149,-268,-170,-229,-181,-241,-235,-260,-225,-272,-259,-355,-270,-344,-221,-429,-254,-394,-305,-443,-348,-406,-390,-405,-383,-420,-382,-385,-345,-366,-308,-315,-295,-317,-301,-307,-307,-319,-316,-337,-273,-310,-260,-332,-260,-287,-238,-267,-235,-236,-259,-223,-218,-231,-226,-221,-205,-245,-216,-236,-203,-244,-208,-224,-221,-211,-210,-180,-186,-192,-188,-204,-183,-150,-161,-144,-182,-151,-199,-152,-176,-159,-170,-168,-142,-147,-118,-110,-117,-97,-125,-84,-116,-88,-125,-124,-122,-160,-117,-177,-157,-199,-179,-171,-156,-148,-174,-137,-139,-117,-130,-135,-112,-124,-104,-131,-80,-123,-82,-99,-80,-101,-46,-83,-75,-90,-80,-63,-59,-51,-122,-65,-82,-66,-92,-66,-83,-92,-88,-117,-122,-158,-126,-150,-152,-142,-139,-83,-139,-101,-146,-84,-121,-95,-110,-91,-88,-79,-72,-47},
+   /* IRC_Composite_C_R0195_T255_P045.wav */
+   {-1,1,-1,1,-1,1,-1,1,-2,2,-2,2,-2,2,-2,2,-3,3,-3,3,-4,4,-4,5,-5,6,-6,7,-8,10,-12,15,-21,35,-138,-127,-91,-119,-131,-100,-136,-154,-123,-162,-225,-17,208,157,422,539,315,-186,-874,-513,1317,2834,3212,2035,1165,1380,1985,1825,760,470,698,708,299,840,390,551,566,771,555,435,745,529,606,365,682,399,459,406,443,291,358,302,320,305,286,249,233,146,69,90,0,-26,-85,-106,-54,-197,-179,-210,-226,-306,-252,-328,-293,-273,-273,-342,-265,-302,-315,-353,-309,-387,-325,-457,-297,-442,-355,-408,-318,-407,-303,-346,-296,-308,-269,-329,-219,-337,-263,-338,-282,-364,-295,-326,-281,-306,-274,-302,-265,-294,-241,-286,-262,-250,-230,-264,-201,-225,-215,-219,-198,-213,-212,-205,-209,-201,-251,-202,-242,-212,-196,-186,-200,-172,-211,-195,-206,-149,-217,-164,-212,-147,-189,-117,-143,-88,-129,-55,-121,-125,-117,-90,-114,-92,-138,-75,-136,-123,-181,-119,-197,-132,-134,-131,-172,-162,-136,-187,-159,-156,-181,-172,-159,-163,-167,-116,-199,-172,-135,-102,-126,-131,-78,-101,-129,-76,-94,-120,-143,-86,-127,-74,-98,-46,-113,-79,-97,-44,-125,-131,-113,-86,-115,-124,-120,-129,-130,-91,-134,-118,-138,-78,-131,-110,-118,-103,-103,-97,-94,-71,-62,-45},
+   /* IRC_Composite_C_R0195_T270_P045.wav */
+   {-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,2,-2,3,-3,4,-4,6,-9,16,-64,-161,-143,-126,-145,-185,-145,-161,-193,-254,-120,-230,-52,146,316,329,329,253,-502,-967,-275,2022,3347,2710,947,1504,1884,1375,1901,809,618,609,671,737,376,610,640,652,766,573,669,526,783,441,547,504,550,598,531,667,314,551,275,554,115,507,96,221,103,170,93,44,90,-71,5,-109,-137,-246,-250,-241,-295,-289,-315,-295,-325,-258,-316,-295,-327,-300,-348,-332,-372,-354,-355,-374,-324,-355,-324,-341,-316,-283,-312,-268,-304,-257,-336,-239,-313,-303,-322,-321,-295,-351,-234,-324,-242,-342,-262,-303,-265,-288,-272,-254,-289,-193,-250,-195,-234,-188,-219,-212,-194,-217,-186,-231,-179,-233,-211,-209,-215,-227,-234,-228,-223,-193,-202,-196,-175,-156,-144,-145,-108,-127,-113,-111,-50,-105,-58,-119,-95,-117,-140,-166,-165,-130,-150,-114,-120,-117,-146,-129,-172,-175,-166,-189,-168,-160,-197,-193,-208,-220,-186,-186,-158,-220,-160,-145,-123,-177,-155,-188,-132,-141,-109,-129,-71,-127,-88,-107,-92,-141,-124,-132,-136,-104,-99,-93,-109,-90,-81,-97,-101,-102,-111,-126,-107,-115,-121,-123,-88,-123,-117,-104,-75,-79,-90,-92,-70,-77,-68,-59},
+   /* IRC_Composite_C_R0195_T285_P045.wav */
+   {5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,6,-6,6,-7,8,-12,29,101,50,50,117,188,58,104,258,239,166,306,686,455,833,834,700,494,424,49,743,2627,3265,3156,1306,-112,1114,559,356,246,113,-51,-113,695,8,344,126,384,-46,304,134,58,125,-42,60,-117,150,-115,60,-66,-74,-290,-174,-222,-336,-410,-393,-454,-500,-436,-457,-579,-566,-638,-618,-691,-652,-777,-695,-736,-701,-743,-633,-655,-650,-600,-573,-602,-585,-524,-544,-563,-496,-527,-483,-513,-414,-513,-387,-442,-351,-441,-335,-395,-360,-389,-346,-380,-350,-337,-339,-350,-286,-303,-303,-289,-278,-317,-289,-279,-256,-274,-224,-223,-201,-206,-165,-204,-151,-205,-147,-195,-148,-200,-177,-178,-170,-179,-162,-156,-149,-146,-135,-109,-105,-142,-94,-76,-93,-61,-56,-26,-52,-21,-71,-105,-113,-130,-140,-154,-74,-110,-107,-129,-114,-156,-108,-147,-143,-197,-121,-103,-164,-177,-160,-133,-135,-137,-138,-104,-105,-75,-105,-83,-126,-87,-67,-44,-97,-18,-59,-13,-45,-23,-53,-56,-43,-12,-14,-6,-40,1,-18,27,-19,35,-53,6,-9,7,11,2,-8,-3,22,16,-2,32,44,18,-5,28,17,14,19,9,27,26,56,44,54,64,65,88},
+   /* IRC_Composite_C_R0195_T300_P045.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-2,6,-170,-102,-259,-234,-199,-268,-51,-528,-137,-354,-180,92,-180,340,604,11,-371,-583,-984,10,2699,3142,3080,1412,327,1987,956,830,764,347,196,382,1091,294,859,889,945,523,1199,597,582,854,625,694,469,918,446,675,763,646,482,552,534,540,563,394,300,434,396,321,144,359,-94,142,-43,165,-138,6,-183,-179,-221,-161,-328,-285,-267,-402,-333,-327,-294,-327,-363,-296,-325,-259,-291,-320,-304,-304,-298,-291,-297,-314,-280,-329,-297,-314,-281,-234,-352,-241,-283,-217,-315,-280,-263,-279,-310,-268,-299,-296,-261,-277,-280,-277,-225,-216,-261,-203,-214,-192,-239,-227,-233,-213,-266,-208,-195,-200,-197,-191,-188,-186,-205,-150,-205,-153,-112,-151,-87,-35,-120,-124,-179,-137,-157,-174,-117,-144,-171,-153,-152,-138,-131,-174,-135,-128,-150,-114,-214,-174,-206,-228,-215,-210,-194,-196,-208,-219,-189,-187,-156,-164,-222,-168,-262,-148,-197,-176,-237,-214,-212,-100,-129,-146,-213,-165,-123,-124,-133,-162,-159,-151,-116,-118,-122,-155,-138,-137,-123,-99,-108,-104,-101,-84,-78,-75,-64,-83,-84,-92,-41,-94,-60,-107,-53,-86,-49,-66,-60,-68,-31,-65,-31,-29},
+   /* IRC_Composite_C_R0195_T315_P045.wav */
+   {-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,1,-1,2,-2,4,-7,17,-100,-204,-340,-371,-160,-344,-275,-415,-419,-249,41,-182,77,891,131,2,-1052,-1535,924,162,4003,4715,1544,1179,621,1424,674,1203,-197,55,273,545,402,710,753,673,897,715,647,832,758,686,732,814,620,688,779,690,604,649,586,620,586,617,478,512,430,467,385,389,275,162,250,131,159,97,100,-42,-92,-107,-156,-193,-270,-280,-274,-295,-312,-336,-301,-367,-310,-306,-332,-327,-275,-299,-308,-302,-264,-320,-240,-345,-250,-325,-245,-353,-231,-309,-234,-308,-240,-335,-236,-284,-266,-330,-242,-316,-259,-281,-275,-309,-222,-254,-279,-277,-236,-254,-242,-235,-247,-199,-188,-240,-203,-141,-139,-214,-131,-200,-170,-111,-70,-193,-145,-148,-101,-173,-117,-218,-163,-214,-197,-208,-111,-206,-113,-191,-104,-100,-70,-142,-89,-169,-106,-184,-156,-187,-185,-255,-195,-151,-192,-223,-180,-149,-215,-245,-216,-248,-232,-230,-274,-292,-203,-223,-171,-218,-208,-215,-141,-139,-194,-204,-163,-184,-155,-159,-154,-164,-134,-151,-130,-134,-130,-152,-122,-129,-133,-120,-106,-118,-102,-72,-84,-69,-84,-65,-68,-55,-55,-56,-72,-45,-57,-43,-50,-64,-34,-44,-40,-36,-22,-14,-32,2},
+   /* IRC_Composite_C_R0195_T330_P045.wav */
+   {0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,-1,1,-1,1,-1,1,-2,2,-2,3,-4,5,-8,-246,-348,-392,-294,-392,-112,-819,-204,-558,205,-381,275,768,476,319,-1186,-2693,1397,173,2904,7126,2191,1840,-141,1440,643,1148,280,-721,280,227,474,413,704,690,753,889,309,920,476,802,503,682,628,623,869,602,783,626,695,707,691,850,635,757,447,558,418,394,227,133,146,217,152,239,143,20,23,13,-113,-130,-110,-235,-231,-231,-223,-290,-311,-279,-339,-297,-335,-258,-358,-326,-332,-275,-283,-293,-287,-240,-279,-296,-244,-248,-261,-257,-280,-251,-271,-271,-291,-272,-276,-292,-293,-287,-273,-297,-261,-259,-291,-282,-238,-254,-285,-269,-242,-239,-234,-237,-211,-232,-160,-167,-153,-54,-121,-133,-68,-127,-55,-93,-170,-253,-307,-209,-255,-315,-307,-329,-320,-102,-172,-106,-41,-45,45,-125,12,-25,-98,-177,-185,-220,-166,-179,-213,-164,-210,-152,-240,-217,-183,-300,-182,-230,-224,-316,-238,-276,-217,-287,-209,-290,-207,-191,-176,-237,-198,-251,-191,-184,-201,-245,-147,-177,-164,-190,-112,-183,-108,-171,-100,-163,-85,-153,-101,-137,-74,-146,-57,-120,-58,-110,-43,-90,-50,-94,-38,-79,-64,-64,-54,-43,-59,-47,-30,-23,-30,-38,-10,-18,12,-13,10,-24,5},
+   /* IRC_Composite_C_R0195_T345_P045.wav */
+   {16,-16,16,-16,16,-16,16,-16,16,-15,15,-14,14,-13,11,-9,7,-4,0,5,-13,25,-45,84,-233,-338,-363,-485,-200,-447,-145,-898,-330,-612,95,-690,294,625,1287,-151,69,-4722,1521,-1345,1556,10396,3052,2742,68,1282,287,1583,813,-614,-309,335,28,768,552,558,584,902,480,734,438,738,500,576,491,546,493,681,435,611,561,752,683,864,812,738,819,537,703,581,367,218,0,254,20,344,-127,37,-166,211,-28,-62,-54,-160,-27,-117,-166,-287,-249,-103,-237,-262,-426,-232,-222,-305,-355,-320,-241,-298,-264,-311,-247,-264,-248,-253,-288,-211,-240,-212,-321,-248,-261,-248,-306,-335,-270,-295,-235,-360,-279,-274,-198,-287,-320,-253,-257,-194,-270,-236,-243,-201,-255,-169,-230,-203,-38,-204,-189,64,-47,-32,-96,-24,-149,-249,-211,-363,-445,-322,-199,-451,-374,-212,-206,-56,-101,52,-41,90,144,4,-84,38,-111,-234,-118,-217,-227,-246,-157,-274,-235,-253,-192,-341,-200,-287,-253,-258,-299,-242,-317,-247,-273,-208,-264,-246,-273,-176,-209,-205,-204,-238,-186,-184,-158,-218,-216,-193,-143,-174,-141,-179,-144,-152,-78,-161,-124,-122,-54,-126,-102,-95,-62,-64,-102,-54,-75,-51,-32,-78,-64,-53,-39,-81,-33,-75,-14,-32,-27,-68,2,-35,2,-22,5,-24,24,-12,-6,0}};
+
+const int16_t irc_composite_c_r0195_p060[][256] =
+  {/* IRC_Composite_C_R0195_T000_P060.wav */
+   {6,-6,6,-6,6,-6,6,-6,6,-6,6,-6,6,-6,5,-5,5,-6,7,-9,18,-62,-477,-304,-336,-458,-335,-491,-402,-829,-409,-707,-293,-66,983,571,504,539,-3665,-1218,-4247,2405,12687,3792,3387,222,562,745,2239,1323,-741,-286,-83,813,783,665,-6,1300,432,669,373,824,513,741,332,740,326,679,126,625,66,777,213,737,590,515,621,819,662,597,612,571,221,378,180,456,-17,193,-19,201,-103,188,-114,-50,-223,41,-272,-7,-204,-73,-274,-27,-179,-118,-266,-52,-236,-150,-298,-185,-328,-211,-330,-359,-321,-288,-288,-390,-233,-301,-238,-287,-186,-257,-179,-231,-185,-249,-207,-244,-242,-257,-261,-239,-326,-213,-312,-252,-311,-214,-308,-230,-270,-210,-269,-227,-231,-256,-281,-231,-249,-231,-210,-238,-174,-214,-204,-218,-156,-225,-179,-158,-114,-151,-171,-128,-150,-184,-87,-97,-145,-171,-201,-199,-179,-259,-239,-309,-310,-314,-227,-196,-170,-192,-26,-115,43,-61,74,-180,-52,-139,-83,-192,-18,-187,-104,-111,-67,-175,-118,-149,-164,-160,-118,-201,-219,-161,-187,-241,-189,-199,-195,-205,-163,-179,-154,-122,-150,-171,-167,-152,-190,-183,-192,-193,-159,-180,-150,-160,-87,-151,-131,-124,-50,-117,-92,-81,-58,-70,-16,-38,-34,-12,39,-10,47,38,87,17,83,31,82,15,52,1},
+   /* IRC_Composite_C_R0195_T030_P060.wav */
+   {20,-21,22,-23,24,-25,26,-28,29,-31,33,-35,37,-40,44,-49,57,-69,99,-268,-386,-296,-341,-306,-439,-461,-219,-926,-450,-515,-185,-151,2064,-248,2208,-557,-2551,-2664,-6482,6173,13499,9000,-215,137,1397,897,1847,1331,-989,-1152,-31,458,894,968,314,922,758,306,312,862,265,458,604,466,114,555,424,349,321,475,497,583,497,442,464,481,505,592,441,239,76,20,163,45,-273,-262,-336,-166,-64,-152,-189,-166,-140,-223,-230,-243,-348,-277,-328,-243,-252,-223,-247,-223,-225,-225,-211,-310,-229,-259,-253,-320,-259,-356,-330,-309,-289,-302,-271,-256,-257,-245,-269,-281,-283,-267,-256,-268,-267,-280,-263,-261,-261,-284,-289,-275,-272,-247,-274,-279,-271,-276,-305,-292,-268,-282,-266,-248,-248,-229,-247,-221,-202,-217,-233,-213,-204,-239,-209,-216,-194,-227,-155,-201,-185,-243,-159,-264,-210,-189,-201,-282,-175,-151,-178,-186,-73,-151,-201,-182,-71,-254,-132,-94,-52,-235,-22,-27,-101,-159,-17,-130,-166,-55,-23,-206,-132,-126,-159,-211,-63,-187,-189,-148,-84,-174,-142,-124,-126,-185,-149,-155,-156,-187,-149,-205,-136,-153,-105,-164,-138,-137,-75,-158,-128,-135,-115,-148,-94,-125,-133,-123,-51,-120,-102,-85,-18,-84,-52,-42,8,-51,10,0,9,-20,40,16,14,24,69,21,25,23,33,2},
+   /* IRC_Composite_C_R0195_T060_P060.wav */
+   {49,-49,50,-50,49,-49,48,-47,45,-43,39,-34,27,-17,1,24,-67,158,-573,27,-273,-533,-199,-358,-57,-1237,277,-801,-585,81,1678,-52,2584,-635,2281,-6550,-1907,-4733,12287,16138,2134,560,-1141,3390,1047,2071,-850,-288,-1730,541,499,654,282,1081,13,1053,-59,982,296,639,439,387,97,735,-102,457,298,541,229,951,144,642,-123,725,239,371,-407,173,-217,5,-418,64,-709,-238,-708,-195,-402,-221,-414,-199,-238,-175,-279,-243,-447,-227,-531,-270,-429,-178,-455,-173,-350,-170,-312,-238,-284,-268,-355,-248,-402,-300,-358,-251,-472,-244,-350,-274,-353,-195,-309,-279,-276,-243,-232,-248,-185,-198,-344,-177,-263,-274,-355,-198,-341,-222,-358,-178,-344,-238,-289,-271,-338,-286,-218,-316,-206,-256,-230,-264,-214,-239,-277,-261,-259,-225,-268,-221,-134,-246,-182,-192,-154,-260,-116,-218,-234,-234,-227,-254,-254,-224,-164,-220,-284,-56,-257,-45,-289,-17,-299,22,-225,103,-264,-36,-33,-158,-37,-147,39,-238,-62,-117,-69,-262,42,-153,-162,-86,-22,-214,-31,-193,-19,-225,-87,-128,-143,-84,-102,-114,-95,-106,-146,-143,-5,-308,-86,-140,-102,-254,41,-68,-208,37,-26,-47,-124,31,-99,-131,-35,-104,-59,-141,32,-115,3,-48,1,-54,32,-67,-4,-31,-40,-50,45,-30,-17,43,3,8,10,37},
+   /* IRC_Composite_C_R0195_T090_P060.wav */
+   {20,-22,23,-25,26,-28,30,-33,35,-38,41,-45,49,-54,60,-66,75,-87,-10,-413,-67,-233,-277,-289,-247,-282,-617,-150,-183,967,613,1964,132,3132,-4890,-30,-7492,6867,13211,9263,1898,-2424,3417,1010,3400,-250,-877,-1856,320,-121,560,520,754,353,491,-65,931,362,494,135,792,604,460,-39,664,258,476,82,671,151,141,64,562,-256,-248,-550,-170,-357,-396,-602,-158,-390,-415,-610,-298,-495,-302,-712,-205,-411,-237,-411,-197,-452,-250,-501,-239,-421,-284,-481,-170,-438,-181,-420,-184,-419,-262,-417,-218,-376,-336,-394,-333,-356,-335,-390,-304,-355,-247,-352,-237,-395,-250,-384,-156,-227,-147,-266,-148,-197,-207,-288,-280,-286,-312,-317,-266,-316,-238,-306,-223,-320,-187,-342,-177,-339,-198,-318,-212,-319,-219,-304,-241,-273,-207,-279,-190,-248,-107,-255,-126,-251,-112,-272,-95,-249,-163,-250,-133,-247,-177,-254,-198,-235,-194,-229,-212,-194,-191,-142,-191,-92,-153,-45,-145,-65,-138,-64,-93,-82,-65,-113,-49,-140,-78,-118,-32,-137,-37,-107,-30,-59,-142,-106,-132,-11,-240,49,-159,20,-179,55,-117,-17,-81,-27,-68,-101,-72,-98,-48,-107,-136,-19,-93,-74,-119,94,-211,28,-129,49,-157,78,-96,21,-143,105,-58,-15,-28,13,-42,14,-40,-16,-41,-38,-39,21,-40,-15,-19,-23,-62,3},
+   /* IRC_Composite_C_R0195_T120_P060.wav */
+   {-11,11,-11,11,-11,10,-10,10,-9,9,-8,7,-6,4,-1,-4,11,-23,54,-588,228,-347,10,-393,78,-279,-133,-619,373,107,880,1033,1066,2024,-241,-1833,-4036,1812,3742,13338,5090,1325,-1146,2733,2814,1516,-118,-2246,-488,96,262,192,815,13,476,529,495,181,475,265,780,210,341,81,498,784,541,47,-30,-8,36,-29,-17,-592,-609,-210,-251,-604,-616,-485,-190,-407,-234,-662,-290,-550,-301,-639,-361,-458,-250,-462,-279,-431,-205,-390,-287,-423,-271,-396,-265,-443,-263,-432,-176,-477,-278,-447,-188,-475,-218,-453,-216,-462,-223,-420,-282,-431,-248,-385,-293,-403,-276,-425,-204,-367,-74,-400,-116,-375,-65,-372,-120,-338,-171,-350,-192,-321,-226,-351,-196,-350,-234,-344,-164,-341,-158,-306,-138,-338,-156,-268,-116,-265,-176,-243,-204,-212,-198,-229,-226,-192,-144,-137,-153,-181,-152,-194,-135,-178,-195,-187,-227,-175,-208,-170,-244,-159,-254,-137,-245,-163,-226,-152,-162,-107,-156,-107,-123,-75,-95,-2,-107,-23,-62,-12,-74,-60,-54,-88,-41,-86,-11,-107,-13,-74,-29,-98,-23,-52,-89,-64,-61,-71,-89,-73,19,-96,13,-86,41,-106,92,-5,1,-25,-7,-37,-56,34,-91,-24,-86,10,-49,-33,-31,-56,10,-83,56,-62,45,-72,47,-53,7,-38,12,-40,-11,-4,-41,-34,-33,-33},
+   /* IRC_Composite_C_R0195_T150_P060.wav */
+   {-16,16,-17,17,-18,18,-19,20,-20,21,-21,22,-22,23,-23,23,-23,22,-20,15,-1,-129,61,-83,-118,-17,-51,-30,-188,-95,109,510,730,1005,940,2137,-1039,-410,-2971,1165,5406,9365,3937,2137,-649,991,4216,933,-931,-1885,270,39,471,-2,92,175,614,97,357,338,829,636,315,-32,510,435,434,61,257,-175,-293,-260,-101,-474,-492,-360,-265,-315,-367,-291,-300,-567,-400,-366,-215,-495,-451,-522,-369,-523,-275,-402,-310,-403,-263,-346,-267,-319,-256,-370,-300,-367,-298,-401,-295,-443,-304,-417,-322,-444,-298,-381,-312,-439,-329,-390,-317,-384,-301,-362,-302,-383,-295,-368,-322,-392,-272,-324,-246,-326,-211,-303,-225,-323,-240,-326,-214,-295,-235,-325,-259,-307,-231,-265,-204,-266,-221,-283,-193,-226,-150,-215,-156,-235,-140,-226,-173,-226,-138,-181,-139,-211,-149,-228,-146,-230,-122,-202,-104,-157,-89,-169,-122,-180,-141,-172,-119,-242,-182,-247,-146,-221,-160,-209,-160,-222,-134,-194,-158,-172,-105,-115,-90,-104,-68,-78,-70,-76,-8,-32,19,-27,16,-5,-14,-32,0,-60,-11,-71,10,-79,-60,-81,-34,-99,-28,-10,-16,0,23,9,-66,-19,-7,-4,-45,41,7,-5,-25,-4,7,-31,-3,22,-19,-42,6,-62,-7,-36,56,-32,47,-21,39,-84,-6,-47,-14,-45,33,-19,-3,-57},
+   /* IRC_Composite_C_R0195_T180_P060.wav */
+   {10,-10,10,-11,11,-10,10,-10,10,-10,10,-10,9,-9,8,-7,6,-5,3,-1,-2,6,-12,19,-22,153,-20,-54,110,44,10,-80,172,36,605,639,998,719,1758,-623,-249,-1810,1268,4590,6942,3540,1131,148,816,4013,533,-755,-1438,271,-13,605,142,55,202,291,-131,465,665,844,217,120,213,663,92,-57,-100,67,-191,-270,-528,-381,-572,-177,-319,-6,-278,-198,-211,58,-313,-463,-592,-248,-465,-368,-496,-344,-514,-270,-361,-242,-341,-263,-349,-193,-335,-278,-416,-272,-416,-335,-439,-313,-451,-348,-453,-349,-443,-392,-472,-363,-430,-386,-408,-312,-392,-320,-379,-310,-370,-331,-403,-335,-385,-348,-366,-269,-311,-276,-302,-234,-290,-316,-341,-275,-241,-217,-208,-227,-228,-238,-222,-184,-224,-208,-241,-186,-232,-205,-245,-203,-214,-163,-151,-128,-135,-145,-180,-187,-204,-171,-188,-145,-158,-112,-157,-75,-116,-69,-149,-129,-160,-136,-115,-133,-152,-184,-177,-167,-165,-197,-218,-169,-194,-138,-205,-133,-162,-112,-132,-87,-117,-57,-79,-38,-55,-39,-50,-29,-58,-18,2,13,-3,31,17,9,8,29,-11,-10,-3,19,-49,-35,-14,-39,-21,-64,-32,-53,-24,-15,-14,-23,27,17,51,-9,-15,-18,18,-37,5,-74,-13,-18,63,0,46,7,63,-15,19,-10,7,-30,-32,-44,-79,-107},
+   /* IRC_Composite_C_R0195_T210_P060.wav */
+   {-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-5,5,-4,4,-4,4,-4,-36,-70,2,-43,-45,-73,17,-83,-76,-100,274,192,709,499,1049,474,192,-1587,-214,1449,4357,5043,2092,784,976,2790,1402,1131,-633,70,310,385,217,300,87,393,309,640,644,573,357,469,403,414,393,291,95,9,177,41,-38,-147,-145,-178,-91,51,-23,82,6,81,-66,-34,-205,-207,-298,-223,-348,-275,-305,-257,-377,-308,-322,-230,-324,-269,-300,-305,-334,-286,-346,-334,-384,-351,-411,-405,-389,-413,-415,-387,-382,-401,-384,-393,-405,-388,-417,-364,-399,-357,-398,-374,-392,-376,-343,-359,-316,-314,-243,-324,-280,-326,-283,-293,-262,-262,-275,-249,-231,-206,-195,-215,-224,-241,-210,-220,-242,-276,-255,-264,-198,-192,-190,-189,-210,-164,-172,-169,-172,-189,-153,-193,-127,-166,-117,-158,-112,-128,-133,-107,-136,-117,-161,-108,-110,-142,-127,-184,-138,-172,-147,-192,-188,-193,-158,-153,-183,-168,-189,-142,-164,-152,-151,-135,-105,-91,-78,-46,-70,-9,-22,-10,6,-3,11,9,0,-18,-34,-49,-47,-78,-56,-80,-88,-84,-41,-76,-89,-89,-34,-62,21,-11,29,-63,38,13,32,-26,-6,-18,-15,-14,-11,-43,-37,-18,40,-10,3,-52,3,-65,-18,-66,-33,-83,-53},
+   /* IRC_Composite_C_R0195_T240_P060.wav */
+   {-5,5,-5,5,-5,6,-6,6,-6,6,-6,6,-6,6,-6,6,-6,6,-6,6,-6,7,-7,7,-6,6,-6,4,1,-68,-141,-43,-120,-98,-71,-122,-111,-111,-115,-151,-61,253,299,522,567,502,242,-1198,-647,857,3187,4265,2111,1152,1068,2588,1288,1124,171,189,245,613,495,306,631,410,664,513,726,614,455,361,493,538,415,305,397,398,328,162,355,127,167,95,91,92,127,159,137,126,136,112,42,-54,-59,-214,-99,-334,-210,-340,-229,-357,-264,-345,-232,-312,-253,-294,-304,-352,-288,-347,-349,-364,-352,-373,-394,-376,-368,-434,-376,-402,-393,-434,-406,-412,-357,-430,-351,-354,-374,-337,-289,-323,-276,-287,-233,-304,-270,-311,-283,-321,-267,-321,-256,-305,-231,-272,-202,-256,-217,-272,-229,-238,-234,-247,-261,-261,-253,-218,-198,-218,-190,-226,-148,-209,-150,-201,-174,-181,-174,-172,-158,-155,-141,-131,-122,-138,-117,-118,-119,-130,-144,-143,-130,-146,-135,-155,-149,-172,-147,-158,-167,-175,-169,-144,-176,-150,-108,-129,-129,-136,-93,-143,-97,-99,-87,-71,-74,-15,-38,-48,-75,-66,-95,-101,-91,-93,-103,-154,-122,-128,-116,-84,-70,-79,-65,-51,-28,-23,-20,-21,-53,-15,-9,15,-49,5,-40,-18,-42,-33,-30,-86,-81,-68,-62,-85,-44,-26,-89,-49,-55,-24,-61,-16},
+   /* IRC_Composite_C_R0195_T270_P060.wav */
+   {5,-5,6,-6,6,-6,6,-6,7,-7,7,-7,8,-8,8,-9,9,-10,10,-11,12,-12,13,-15,16,-19,22,-27,38,-79,-217,-116,-99,-179,-211,-36,-231,-204,-130,-219,-231,-133,80,277,396,430,429,-88,-768,-1080,327,3345,4354,1751,849,1821,1217,1828,885,517,269,311,1017,365,820,357,716,549,590,621,516,736,326,780,263,770,281,705,378,636,371,475,311,429,255,409,238,273,140,314,213,208,138,105,115,-23,74,-205,-25,-248,-155,-287,-280,-252,-297,-307,-288,-293,-267,-333,-245,-352,-304,-348,-348,-351,-433,-337,-451,-334,-472,-324,-423,-375,-381,-372,-314,-375,-276,-267,-321,-301,-279,-273,-291,-287,-302,-304,-304,-290,-236,-296,-289,-263,-294,-284,-324,-265,-281,-292,-245,-245,-266,-260,-214,-224,-254,-200,-260,-209,-252,-220,-200,-254,-196,-233,-158,-228,-136,-185,-148,-205,-143,-175,-148,-149,-166,-120,-148,-104,-128,-140,-125,-152,-101,-130,-77,-199,-88,-178,-108,-197,-126,-198,-117,-194,-113,-172,-133,-128,-87,-131,-57,-112,-60,-136,-71,-150,-125,-162,-106,-205,-139,-165,-138,-149,-118,-134,-84,-134,-83,-120,-64,-109,-79,-116,-53,-90,-64,-49,-97,-69,-54,-86,-38,-63,-120,-64,-43,-89,-78,-45,-32,-110,-9,-49,-30,-91,-36,-87,-34,-90,-12,-62,-20,-67},
+   /* IRC_Composite_C_R0195_T300_P060.wav */
+   {5,-5,5,-5,6,-6,6,-6,6,-6,6,-6,6,-7,7,-7,7,-7,7,-8,8,-8,8,-9,9,-9,10,-11,16,-103,-316,-221,-172,-255,-229,-364,-48,-533,-236,-67,-300,259,642,185,100,427,-1196,-1745,476,2467,5309,2589,680,1682,925,1569,1041,949,-237,234,798,421,371,569,750,604,969,544,826,529,786,297,628,582,490,514,782,585,536,518,533,446,484,585,264,478,477,367,328,417,65,219,65,246,-91,156,-64,72,-138,50,-297,-48,-254,-210,-356,-206,-256,-287,-317,-214,-367,-254,-373,-339,-401,-317,-430,-376,-368,-350,-418,-316,-298,-353,-273,-305,-281,-287,-227,-371,-293,-252,-310,-323,-260,-272,-324,-243,-257,-310,-290,-276,-297,-255,-201,-324,-280,-270,-230,-282,-221,-248,-298,-222,-268,-241,-260,-255,-277,-235,-187,-236,-210,-172,-224,-173,-145,-191,-157,-155,-157,-126,-186,-84,-150,-177,-153,-72,-128,-133,-127,-100,-229,-197,-160,-241,-236,-171,-176,-120,-75,-47,-120,-105,-128,-76,-160,-120,-127,-109,-139,-135,-144,-192,-179,-152,-203,-173,-143,-161,-61,-179,-143,-104,-126,-147,-78,-176,-138,-177,-87,-160,-110,-137,-186,-155,-86,-135,-127,-105,-97,-124,-34,-111,-50,-117,-72,-123,-75,-65,-94,-51,-83,-75,-56,-47,-45,-13,-39,-55,20,-56,36,-41,12,-24},
+   /* IRC_Composite_C_R0195_T330_P060.wav */
+   {3,-3,4,-4,4,-4,5,-5,5,-6,6,-6,7,-7,8,-9,10,-11,13,-15,18,-22,28,-41,71,-233,-373,-96,-345,-318,-371,-265,-252,-594,-439,-312,-197,197,323,301,1318,-757,-1616,-868,-2081,2422,6337,5378,1528,381,1195,1007,1545,810,158,-165,224,315,652,555,370,728,945,600,540,730,606,560,567,567,493,662,473,574,547,639,468,638,527,625,475,584,476,643,403,465,284,372,243,235,43,108,49,102,20,-28,-86,-48,-49,-120,-149,-133,-123,-173,-198,-229,-246,-291,-242,-300,-328,-373,-296,-357,-299,-377,-351,-356,-276,-311,-324,-289,-282,-268,-271,-255,-256,-273,-249,-285,-230,-270,-219,-299,-282,-269,-275,-287,-307,-267,-290,-268,-297,-284,-263,-248,-257,-244,-243,-257,-290,-273,-259,-265,-237,-261,-238,-273,-209,-234,-171,-171,-157,-118,-120,-137,-80,-115,-96,-166,-45,-140,-79,-126,-114,-259,-243,-265,-228,-274,-267,-264,-220,-203,-117,-137,-118,-112,-33,-82,-41,-93,-102,-121,-120,-145,-175,-177,-147,-143,-65,-127,-138,-136,-94,-127,-130,-217,-183,-159,-168,-202,-184,-203,-240,-202,-146,-195,-160,-147,-169,-143,-72,-128,-191,-139,-128,-170,-166,-129,-160,-152,-97,-112,-123,-68,-72,-87,-76,-27,-63,-48,-50,-16,-43,16,-15,-10,-1,57,47,30,49,50,35,14}};
+
+const int16_t irc_composite_c_r0195_p075[][256] =
+  {/* IRC_Composite_C_R0195_T000_P075.wav */
+   {12,-12,13,-13,13,-13,13,-13,14,-14,14,-14,14,-14,15,-15,14,-14,13,-12,7,8,-224,-367,-189,-595,-344,-165,-801,-581,-302,-836,-618,407,530,134,1455,-966,-265,-4329,-4113,4525,6992,8542,304,-199,2762,1323,1544,1359,195,-1272,815,548,454,489,1188,303,1217,433,606,486,940,200,865,394,415,452,439,167,687,262,388,668,618,250,776,505,518,525,594,287,552,461,358,276,304,160,239,30,35,-21,-62,-125,-17,-31,-371,39,-211,-122,-197,40,-391,72,-166,-93,-262,-84,-337,-76,-325,-174,-301,-196,-378,-120,-367,-324,-293,-268,-402,-208,-271,-377,-322,-212,-314,-262,-200,-245,-294,-186,-222,-275,-226,-280,-246,-300,-262,-182,-254,-272,-206,-217,-307,-182,-214,-248,-250,-184,-258,-303,-196,-229,-312,-240,-213,-256,-268,-167,-202,-256,-145,-174,-141,-306,-39,-235,-192,-132,-161,-142,-205,-78,-192,-93,-268,-65,-239,-166,-229,-88,-215,-227,-134,-262,-99,-230,-94,-297,-147,-247,-15,-245,-197,-51,-171,-94,-46,-60,-97,-14,-100,14,-99,-80,-92,-24,-178,55,-112,-23,-104,47,-121,-73,-80,-27,-196,-98,-74,-188,-162,-54,-110,-182,-133,-30,-121,-83,-72,-116,-145,-93,-89,-136,-159,-163,-75,-145,-145,-85,-61,-103,-107,5,-74,-43,-3,-22,-72,-26,-24,6,-85,5},
+   /* IRC_Composite_C_R0195_T060_P075.wav */
+   {-14,15,-15,16,-17,18,-19,20,-21,22,-23,24,-25,27,-28,30,-32,35,-42,65,-281,-389,123,-583,-268,-119,-384,-436,-660,282,-1424,931,-191,1870,-113,1526,243,-2974,-2922,-3038,6766,14120,3000,-177,427,2482,2472,607,1128,-2092,683,-982,2038,-839,1456,-30,1050,188,720,669,391,864,257,483,214,493,238,321,378,114,592,45,886,-9,646,103,701,13,345,57,116,121,-88,171,-386,97,-335,-262,-369,-373,-308,-409,33,-533,5,-484,24,-435,-237,-319,-208,-233,-416,-146,-317,-166,-309,-232,-304,-340,-158,-439,-169,-482,-113,-521,-200,-450,-258,-409,-296,-298,-425,-296,-402,-280,-434,-185,-317,-227,-279,-194,-241,-311,-232,-320,-314,-306,-233,-329,-276,-236,-245,-297,-256,-187,-267,-262,-210,-246,-287,-248,-205,-331,-278,-229,-232,-260,-191,-174,-217,-208,-141,-202,-243,-151,-110,-284,-132,-132,-134,-277,-115,-207,-229,-254,-153,-213,-329,-111,-236,-248,-245,-82,-281,-102,-144,-93,-133,-52,-82,-139,-23,-121,-40,-135,7,-127,-34,-127,-9,-67,-119,-37,-34,-134,4,-70,-77,-81,-62,-64,-127,-18,-109,11,-156,60,-224,27,-77,-130,6,-101,-38,-151,93,-175,-123,0,-32,-129,-111,115,-229,-48,12,-46,-196,-37,-14,-163,-83,-1,-137,-48,-101,91,-141,15,-60,28,-35,-33,-10,-28},
+   /* IRC_Composite_C_R0195_T120_P075.wav */
+   {-13,13,-14,14,-15,15,-16,16,-17,17,-18,18,-19,19,-20,20,-19,19,-16,12,-1,-57,-48,-189,-132,-75,-227,-56,-343,28,-501,199,201,917,964,1130,545,1226,-3782,-981,-1166,8252,10379,3341,-521,-669,4572,428,2096,-1386,-405,-1098,1383,-2,236,525,510,124,361,340,604,403,459,20,703,550,361,4,507,76,379,200,87,66,-36,226,133,30,-418,-183,-144,-119,-262,-333,-294,-550,-263,-353,-297,-518,-525,-481,-405,-172,-382,-282,-383,-195,-329,-205,-358,-340,-304,-296,-417,-305,-321,-319,-400,-254,-454,-283,-418,-294,-482,-262,-427,-329,-482,-271,-419,-319,-403,-345,-351,-373,-409,-381,-310,-297,-225,-271,-290,-302,-223,-252,-250,-249,-220,-293,-267,-259,-270,-267,-281,-238,-321,-208,-300,-203,-357,-197,-295,-169,-277,-177,-273,-210,-234,-196,-252,-188,-188,-144,-221,-121,-160,-150,-180,-111,-124,-171,-110,-188,-133,-213,-124,-220,-168,-203,-176,-182,-229,-156,-255,-117,-221,-99,-240,-61,-158,-122,-129,-39,-92,-133,4,-74,-53,-77,35,-129,8,-50,5,-61,-24,-33,-46,31,-56,58,-35,39,-37,57,-74,2,-108,-10,-136,18,-141,8,-114,4,-132,-22,-37,-31,-20,-71,79,-149,14,-32,50,-85,-47,94,-59,21,-39,82,-145,47,-47,-14,-82,-16,-3,-58,-48,-44,-22,-40},
+   /* IRC_Composite_C_R0195_T180_P075.wav */
+   {-17,17,-17,17,-16,16,-16,16,-15,15,-15,14,-14,14,-13,12,-12,11,-10,9,-7,5,-2,-2,11,164,-118,156,-74,171,-166,281,-259,331,-163,647,483,1342,96,2125,-652,369,-2506,1012,3788,6928,5436,-854,927,1011,2973,419,642,-1377,-224,371,309,-128,78,268,228,233,434,427,397,-91,692,541,379,-60,222,-23,131,178,60,-172,-56,-82,-145,-311,-297,-431,-248,-412,-107,-371,-72,-353,-277,-448,-332,-408,-362,-461,-379,-530,-388,-407,-290,-364,-334,-409,-288,-419,-285,-384,-338,-447,-343,-386,-344,-453,-378,-390,-377,-502,-386,-429,-361,-457,-343,-463,-441,-462,-348,-398,-369,-416,-375,-435,-312,-336,-244,-309,-307,-321,-263,-272,-272,-262,-236,-196,-217,-232,-215,-261,-261,-304,-256,-296,-220,-241,-218,-248,-199,-190,-206,-213,-223,-195,-209,-182,-150,-161,-155,-165,-145,-155,-129,-127,-102,-102,-126,-107,-145,-133,-143,-141,-131,-140,-117,-119,-86,-121,-100,-131,-130,-130,-139,-142,-162,-155,-148,-128,-171,-136,-140,-117,-101,-76,-112,-44,-78,-20,-37,-38,-18,15,-6,2,39,44,55,-20,46,41,30,8,37,20,-14,0,-10,0,-57,-38,-55,-62,-113,-27,-88,-33,-69,-21,-80,1,-86,62,-11,16,23,128,6,102,-12,152,3,46,29,36,-97,2,-15,22,-109,-54},
+   /* IRC_Composite_C_R0195_T240_P075.wav */
+   {3,-3,3,-3,3,-3,3,-2,2,-2,2,-2,1,-1,1,0,0,1,-2,3,-4,5,-7,10,-13,20,-42,-111,-129,-71,-115,-137,-61,-159,-123,-192,-19,-219,406,390,680,549,886,-353,-736,-1471,543,4694,6064,1476,785,1081,2341,1848,567,793,-1054,878,-19,694,-27,624,453,574,713,561,524,289,447,341,517,436,227,504,311,411,259,396,204,319,114,96,38,-78,39,-23,9,123,-104,210,-61,44,-245,-60,-225,-163,-284,-292,-366,-364,-318,-284,-326,-284,-257,-269,-279,-348,-246,-376,-367,-329,-388,-355,-428,-316,-418,-375,-426,-318,-396,-436,-393,-407,-383,-405,-383,-442,-422,-393,-344,-375,-309,-340,-270,-311,-270,-269,-256,-268,-233,-283,-233,-230,-236,-283,-256,-265,-301,-274,-317,-261,-298,-265,-239,-262,-221,-248,-188,-287,-192,-249,-152,-246,-140,-231,-148,-218,-130,-170,-165,-154,-171,-127,-181,-99,-138,-109,-120,-99,-88,-129,-61,-150,-63,-199,-44,-206,-106,-201,-113,-189,-182,-146,-180,-174,-177,-146,-150,-167,-103,-131,-76,-138,-14,-119,-32,-66,-17,-34,-48,8,-80,-30,-22,-34,-12,-52,1,-29,-7,12,-34,-30,-36,-43,-95,-29,-80,-102,-17,-119,-24,-119,10,-78,-16,-105,-38,-66,-45,-15,9,-36,-32,33,-12,-9,-66,-12,-59,8,-23,-10,-6,-22},
+   /* IRC_Composite_C_R0195_T300_P075.wav */
+   {5,-5,5,-6,6,-6,6,-7,7,-7,8,-8,9,-9,10,-10,11,-12,13,-14,16,-19,23,-30,49,-174,-216,-220,-266,-146,-276,-310,-167,-398,-257,-446,-119,2,150,904,282,399,-249,-994,-2605,51,5569,5353,3304,-294,1237,2394,748,1912,84,-494,511,465,372,641,761,120,1277,373,764,283,1087,-27,731,558,213,625,490,578,331,755,261,576,308,533,270,531,241,337,345,292,249,254,130,252,95,124,0,-94,22,-124,-166,-120,-222,-195,-188,-102,-288,-138,-247,-177,-285,-135,-368,-230,-298,-370,-301,-355,-365,-394,-255,-460,-300,-338,-376,-356,-289,-388,-412,-303,-449,-224,-403,-214,-327,-319,-233,-280,-268,-283,-281,-308,-227,-300,-251,-210,-330,-195,-309,-196,-294,-197,-342,-211,-310,-249,-268,-272,-295,-243,-275,-237,-249,-206,-274,-172,-248,-191,-211,-154,-258,-145,-194,-183,-134,-143,-154,-131,-111,-163,-86,-129,-136,-111,-159,-135,-153,-128,-194,-126,-204,-111,-239,-106,-205,-215,-163,-175,-185,-155,-144,-130,-163,-92,-105,-142,-4,-111,-109,-19,-118,13,-189,9,-76,-118,-8,-6,-126,-54,46,-138,-64,43,-92,-25,-132,48,-164,-57,-43,-117,-199,-11,-164,-58,-105,-75,-100,-150,-59,-62,-137,-80,-148,-65,-126,-49,-62,-33,-77,-10,-18,-11,-21,-15,-36,-38,-38,9}};
+
+const int16_t irc_composite_c_r0195_p090[][256] =
+  {/* IRC_Composite_C_R0195_T000_P090.wav */
+   {-1,1,-1,1,-1,1,-1,2,-2,2,-3,3,-3,4,-5,5,-7,8,-10,14,-25,-385,-133,-356,-344,-390,-312,-438,-400,-852,-107,-995,-245,-186,1358,-771,1473,28,-1710,-3762,-3025,4119,8200,5927,-785,1558,2250,1837,2098,631,-318,150,-77,1215,-23,792,865,600,981,659,366,950,511,766,502,516,589,299,331,571,105,609,462,485,538,594,470,640,393,432,325,604,259,355,336,454,33,263,182,-24,-124,121,-231,-50,-93,49,-230,-101,25,-196,-123,-21,-160,-161,-132,-136,-155,-268,-121,-146,-311,-112,-328,-201,-220,-280,-214,-304,-233,-227,-250,-242,-334,-256,-283,-393,-267,-285,-332,-331,-224,-295,-279,-171,-151,-361,-199,-239,-227,-317,-168,-211,-370,-182,-150,-343,-204,-192,-224,-319,-203,-137,-357,-146,-223,-237,-276,-162,-201,-262,-186,-209,-147,-267,-46,-252,-162,-146,-182,-199,-143,-218,-204,-108,-250,-163,-90,-241,-137,-125,-122,-272,-111,-188,-211,-160,-160,-164,-201,-231,-114,-175,-180,-212,-172,-181,-129,-130,-77,-150,-114,-39,12,-122,-9,-11,-46,-21,-4,7,-73,-20,-50,-21,-80,-102,2,-103,-66,-67,-8,-41,-119,-26,29,-208,3,-108,-34,-228,114,-196,-84,-28,1,-109,-92,77,-169,-27,48,-12,-79,3,-100,-44,-53,13,-103,-90,-14,-91,-23,-64,-21,-116}};
+
+struct Elevation
+{
+    /**
+     * An array of |count| impulse responses of 256 samples for the left ear.
+     * The impulse responses in each elevation are at equally spaced azimuths
+     * for a full 360 degree revolution, ordered clockwise from in front the
+     * listener.
+     */
+    const int16_t (*azimuths)[256];
+    int count;
+};
+
+/**
+ * irc_composite_c_r0195 is an array with each element containing data for one
+ * elevation.
+ */
+const Elevation irc_composite_c_r0195[] =
+  {{irc_composite_c_r0195_p315, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p315)},
+   {irc_composite_c_r0195_p330, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p330)},
+   {irc_composite_c_r0195_p345, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p345)},
+   {irc_composite_c_r0195_p000, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p000)},
+   {irc_composite_c_r0195_p015, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p015)},
+   {irc_composite_c_r0195_p030, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p030)},
+   {irc_composite_c_r0195_p045, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p045)},
+   {irc_composite_c_r0195_p060, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p060)},
+   {irc_composite_c_r0195_p075, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p075)},
+   {irc_composite_c_r0195_p090, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p090)}};
+
+const int irc_composite_c_r0195_first_elevation = -45; /* degrees */
+const int irc_composite_c_r0195_elevation_interval = 15; /* degrees */
+const int irc_composite_c_r0195_sample_rate = 44100; /* Hz */
diff --git a/dom/media/webaudio/blink/PeriodicWave.cpp b/dom/media/webaudio/blink/PeriodicWave.cpp
new file mode 100644
index 0000000000..3f949207a3
--- /dev/null
+++ b/dom/media/webaudio/blink/PeriodicWave.cpp
@@ -0,0 +1,358 @@
+/*
+ * Copyright (C) 2012 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "PeriodicWave.h"
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include "mozilla/FFTBlock.h"
+
+const unsigned MinPeriodicWaveSize = 4096; // This must be a power of two.
+const unsigned MaxPeriodicWaveSize = 8192; // This must be a power of two.
+const float CentsPerRange = 1200 / 3; // 1/3 Octave.
+
+using namespace mozilla;
+using mozilla::dom::OscillatorType;
+
+namespace WebCore {
+
+already_AddRefed<PeriodicWave>
+PeriodicWave::create(float sampleRate,
+                     const float* real,
+                     const float* imag,
+                     size_t numberOfComponents,
+                     bool disableNormalization)
+{
+    bool isGood = real && imag && numberOfComponents > 0;
+    MOZ_ASSERT(isGood);
+    if (isGood) {
+        RefPtr<PeriodicWave> periodicWave =
+            new PeriodicWave(sampleRate, numberOfComponents,
+                             disableNormalization);
+
+        // Limit the number of components used to those for frequencies below the
+        // Nyquist of the fixed length inverse FFT.
+        size_t halfSize = periodicWave->m_periodicWaveSize / 2;
+        numberOfComponents = std::min(numberOfComponents, halfSize);
+        periodicWave->m_numberOfComponents = numberOfComponents;
+        periodicWave->m_realComponents = new AudioFloatArray(numberOfComponents);
+        periodicWave->m_imagComponents = new AudioFloatArray(numberOfComponents);
+        memcpy(periodicWave->m_realComponents->Elements(), real,
+               numberOfComponents * sizeof(float));
+        memcpy(periodicWave->m_imagComponents->Elements(), imag,
+               numberOfComponents * sizeof(float));
+
+        return periodicWave.forget();
+    }
+    return nullptr;
+}
+
+already_AddRefed<PeriodicWave>
+PeriodicWave::createSine(float sampleRate)
+{
+    RefPtr<PeriodicWave> periodicWave =
+        new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+    periodicWave->generateBasicWaveform(OscillatorType::Sine);
+    return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave>
+PeriodicWave::createSquare(float sampleRate)
+{
+    RefPtr<PeriodicWave> periodicWave =
+        new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+    periodicWave->generateBasicWaveform(OscillatorType::Square);
+    return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave>
+PeriodicWave::createSawtooth(float sampleRate)
+{
+    RefPtr<PeriodicWave> periodicWave =
+        new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+    periodicWave->generateBasicWaveform(OscillatorType::Sawtooth);
+    return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave>
+PeriodicWave::createTriangle(float sampleRate)
+{
+    RefPtr<PeriodicWave> periodicWave =
+        new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+    periodicWave->generateBasicWaveform(OscillatorType::Triangle);
+    return periodicWave.forget();
+}
+
+PeriodicWave::PeriodicWave(float sampleRate, size_t numberOfComponents, bool disableNormalization)
+    : m_sampleRate(sampleRate)
+    , m_centsPerRange(CentsPerRange)
+    , m_maxPartialsInBandLimitedTable(0)
+    , m_normalizationScale(1.0f)
+    , m_disableNormalization(disableNormalization)
+{
+    float nyquist = 0.5 * m_sampleRate;
+
+    if (numberOfComponents <= MinPeriodicWaveSize) {
+        m_periodicWaveSize = MinPeriodicWaveSize;
+    } else {
+        unsigned npow2 = powf(2.0f, floorf(logf(numberOfComponents - 1.0)/logf(2.0f) + 1.0f));
+        m_periodicWaveSize = std::min(MaxPeriodicWaveSize, npow2);
+    }
+
+    m_numberOfRanges = (unsigned)(3.0f*logf(m_periodicWaveSize)/logf(2.0f));
+    m_bandLimitedTables.SetLength(m_numberOfRanges);
+    m_lowestFundamentalFrequency = nyquist / maxNumberOfPartials();
+    m_rateScale = m_periodicWaveSize / m_sampleRate;
+}
+
+size_t PeriodicWave::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    amount += m_bandLimitedTables.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_bandLimitedTables.Length(); i++) {
+        if (m_bandLimitedTables[i]) {
+            amount += m_bandLimitedTables[i]->ShallowSizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    return amount;
+}
+
+void PeriodicWave::waveDataForFundamentalFrequency(float fundamentalFrequency, float* &lowerWaveData, float* &higherWaveData, float& tableInterpolationFactor)
+{
+
+    // Negative frequencies are allowed, in which case we alias
+    // to the positive frequency.
+    fundamentalFrequency = fabsf(fundamentalFrequency);
+
+    // We only need to rebuild to the tables if the new fundamental
+    // frequency is low enough to allow for more partials below the
+    // Nyquist frequency.
+    unsigned numberOfPartials = numberOfPartialsForRange(0);
+    float nyquist = 0.5 * m_sampleRate;
+    if (fundamentalFrequency != 0.0) {
+        numberOfPartials = std::min(numberOfPartials, (unsigned)(nyquist / fundamentalFrequency));
+    }
+    if (numberOfPartials > m_maxPartialsInBandLimitedTable) {
+        for (unsigned rangeIndex = 0; rangeIndex < m_numberOfRanges; ++rangeIndex) {
+            m_bandLimitedTables[rangeIndex] = 0;
+        }
+
+        // We need to create the first table to determine the normalization
+        // constant.
+        createBandLimitedTables(fundamentalFrequency, 0);
+        m_maxPartialsInBandLimitedTable = numberOfPartials;
+    }
+
+    // Calculate the pitch range.
+    float ratio = fundamentalFrequency > 0 ? fundamentalFrequency / m_lowestFundamentalFrequency : 0.5;
+    float centsAboveLowestFrequency = logf(ratio)/logf(2.0f) * 1200;
+
+    // Add one to round-up to the next range just in time to truncate
+    // partials before aliasing occurs.
+    float pitchRange = 1 + centsAboveLowestFrequency / m_centsPerRange;
+
+    pitchRange = std::max(pitchRange, 0.0f);
+    pitchRange = std::min(pitchRange, static_cast<float>(m_numberOfRanges - 1));
+
+    // The words "lower" and "higher" refer to the table data having
+    // the lower and higher numbers of partials. It's a little confusing
+    // since the range index gets larger the more partials we cull out.
+    // So the lower table data will have a larger range index.
+    unsigned rangeIndex1 = static_cast<unsigned>(pitchRange);
+    unsigned rangeIndex2 = rangeIndex1 < m_numberOfRanges - 1 ? rangeIndex1 + 1 : rangeIndex1;
+
+    if (!m_bandLimitedTables[rangeIndex1].get())
+        createBandLimitedTables(fundamentalFrequency, rangeIndex1);
+
+    if (!m_bandLimitedTables[rangeIndex2].get())
+        createBandLimitedTables(fundamentalFrequency, rangeIndex2);
+
+    lowerWaveData = m_bandLimitedTables[rangeIndex2]->Elements();
+    higherWaveData = m_bandLimitedTables[rangeIndex1]->Elements();
+
+    // Ranges from 0 -> 1 to interpolate between lower -> higher.
+    tableInterpolationFactor = rangeIndex2 - pitchRange;
+}
+
+unsigned PeriodicWave::maxNumberOfPartials() const
+{
+    return m_periodicWaveSize / 2;
+}
+
+unsigned PeriodicWave::numberOfPartialsForRange(unsigned rangeIndex) const
+{
+    // Number of cents below nyquist where we cull partials.
+    float centsToCull = rangeIndex * m_centsPerRange;
+
+    // A value from 0 -> 1 representing what fraction of the partials to keep.
+    float cullingScale = pow(2, -centsToCull / 1200);
+
+    // The very top range will have all the partials culled.
+    unsigned numberOfPartials = cullingScale * maxNumberOfPartials();
+
+    return numberOfPartials;
+}
+
+// Convert into time-domain wave buffers.
+// One table is created for each range for non-aliasing playback
+// at different playback rates. Thus, higher ranges have more
+// high-frequency partials culled out.
+void PeriodicWave::createBandLimitedTables(float fundamentalFrequency,
+                                           unsigned rangeIndex)
+{
+    unsigned fftSize = m_periodicWaveSize;
+    unsigned i;
+
+    const float *realData = m_realComponents->Elements();
+    const float *imagData = m_imagComponents->Elements();
+
+    // This FFTBlock is used to cull partials (represented by frequency bins).
+    FFTBlock frame(fftSize);
+
+    // Find the starting bin where we should start culling the aliasing
+    // partials for this pitch range.  We need to clear out the highest
+    // frequencies to band-limit the waveform.
+    unsigned numberOfPartials = numberOfPartialsForRange(rangeIndex);
+    // Also limit to the number of components that are provided.
+    numberOfPartials = std::min(numberOfPartials, m_numberOfComponents - 1);
+
+    // Limit number of partials to those below Nyquist frequency
+    float nyquist = 0.5 * m_sampleRate;
+    if (fundamentalFrequency != 0.0) {
+        numberOfPartials = std::min(numberOfPartials,
+                                    (unsigned)(nyquist / fundamentalFrequency));
+    }
+
+    // Copy from loaded frequency data and generate complex conjugate
+    // because of the way the inverse FFT is defined.
+    // The coefficients of higher partials remain zero, as initialized in
+    // the FFTBlock constructor.
+    for (i = 0; i < numberOfPartials + 1; ++i) {
+        frame.RealData(i) = realData[i];
+        frame.ImagData(i) = -imagData[i];
+    }
+
+    // Clear any DC-offset.
+    frame.RealData(0) = 0;
+    // Clear value which has no effect.
+    frame.ImagData(0) = 0;
+
+    // Create the band-limited table.
+    AlignedAudioFloatArray* table = new AlignedAudioFloatArray(m_periodicWaveSize);
+    m_bandLimitedTables[rangeIndex] = table;
+
+    // Apply an inverse FFT to generate the time-domain table data.
+    float* data = m_bandLimitedTables[rangeIndex]->Elements();
+    frame.GetInverseWithoutScaling(data);
+
+    // For the first range (which has the highest power), calculate
+    // its peak value then compute normalization scale.
+    if (!m_disableNormalization && !rangeIndex) {
+        float maxValue;
+        maxValue = AudioBufferPeakValue(data, m_periodicWaveSize);
+
+        if (maxValue)
+            m_normalizationScale = 1.0f / maxValue;
+    }
+
+    // Apply normalization scale.
+    if (!m_disableNormalization) {
+      AudioBufferInPlaceScale(data, m_normalizationScale, m_periodicWaveSize);
+    }
+}
+
+void PeriodicWave::generateBasicWaveform(OscillatorType shape)
+{
+    const float piFloat = float(M_PI);
+    unsigned fftSize = periodicWaveSize();
+    unsigned halfSize = fftSize / 2;
+
+    m_numberOfComponents = halfSize;
+    m_realComponents = new AudioFloatArray(halfSize);
+    m_imagComponents = new AudioFloatArray(halfSize);
+    float* realP = m_realComponents->Elements();
+    float* imagP = m_imagComponents->Elements();
+
+    // Clear DC and imag value which is ignored.
+    realP[0] = 0;
+    imagP[0] = 0;
+
+    for (unsigned n = 1; n < halfSize; ++n) {
+        float omega = 2 * piFloat * n;
+        float invOmega = 1 / omega;
+
+        // Fourier coefficients according to standard definition.
+        float a; // Coefficient for cos().
+        float b; // Coefficient for sin().
+
+        // Calculate Fourier coefficients depending on the shape.
+        // Note that the overall scaling (magnitude) of the waveforms
+        // is normalized in createBandLimitedTables().
+        switch (shape) {
+        case OscillatorType::Sine:
+            // Standard sine wave function.
+            a = 0;
+            b = (n == 1) ? 1 : 0;
+            break;
+        case OscillatorType::Square:
+            // Square-shaped waveform with the first half its maximum value
+            // and the second half its minimum value.
+            a = 0;
+            b = invOmega * ((n & 1) ? 2 : 0);
+            break;
+        case OscillatorType::Sawtooth:
+            // Sawtooth-shaped waveform with the first half ramping from
+            // zero to maximum and the second half from minimum to zero.
+            a = 0;
+            b = -invOmega * cos(0.5 * omega);
+            break;
+        case OscillatorType::Triangle:
+            // Triangle-shaped waveform going from its maximum value to
+            // its minimum value then back to the maximum value.
+            a = 0;
+            if (n & 1) {
+              b = 2 * (2 / (n * piFloat) * 2 / (n * piFloat)) * ((((n - 1) >> 1) & 1) ? -1 : 1);
+            } else {
+              b = 0;
+            }
+            break;
+        default:
+            NS_NOTREACHED("invalid oscillator type");
+            a = 0;
+            b = 0;
+            break;
+        }
+
+        realP[n] = a;
+        imagP[n] = b;
+    }
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/PeriodicWave.h b/dom/media/webaudio/blink/PeriodicWave.h
new file mode 100644
index 0000000000..47381d450a
--- /dev/null
+++ b/dom/media/webaudio/blink/PeriodicWave.h
@@ -0,0 +1,118 @@
+/*
+ * Copyright (C) 2012 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef PeriodicWave_h
+#define PeriodicWave_h
+
+#include "mozilla/dom/OscillatorNodeBinding.h"
+#include <nsAutoPtr.h>
+#include <nsTArray.h>
+#include "AlignedTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+typedef AlignedTArray<float> AlignedAudioFloatArray;
+typedef nsTArray<float> AudioFloatArray;
+
+class PeriodicWave {
+public:
+    NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebCore::PeriodicWave);
+
+    static already_AddRefed<PeriodicWave> createSine(float sampleRate);
+    static already_AddRefed<PeriodicWave> createSquare(float sampleRate);
+    static already_AddRefed<PeriodicWave> createSawtooth(float sampleRate);
+    static already_AddRefed<PeriodicWave> createTriangle(float sampleRate);
+
+    // Creates an arbitrary periodic wave given the frequency components
+    // (Fourier coefficients).
+    static already_AddRefed<PeriodicWave> create(float sampleRate,
+                                                 const float* real,
+                                                 const float* imag,
+                                                 size_t numberOfComponents,
+                                                 bool disableNormalization);
+
+    // Returns pointers to the lower and higher wave data for the pitch range
+    // containing the given fundamental frequency. These two tables are in
+    // adjacent "pitch" ranges where the higher table will have the maximum
+    // number of partials which won't alias when played back at this
+    // fundamental frequency. The lower wave is the next range containing fewer
+    // partials than the higher wave. Interpolation between these two tables
+    // can be made according to tableInterpolationFactor. Where values
+    // from 0 -> 1 interpolate between lower -> higher.
+    void waveDataForFundamentalFrequency(float, float* &lowerWaveData, float* &higherWaveData, float& tableInterpolationFactor);
+
+    // Returns the scalar multiplier to the oscillator frequency to calculate
+    // wave buffer phase increment.
+    float rateScale() const { return m_rateScale; }
+
+    unsigned periodicWaveSize() const { return m_periodicWaveSize; }
+    float sampleRate() const { return m_sampleRate; }
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    explicit PeriodicWave(float sampleRate, size_t numberOfComponents, bool disableNormalization);
+    ~PeriodicWave() {}
+
+    void generateBasicWaveform(mozilla::dom::OscillatorType);
+
+    float m_sampleRate;
+    unsigned m_periodicWaveSize;
+    unsigned m_numberOfRanges;
+    float m_centsPerRange;
+    unsigned m_numberOfComponents;
+    nsAutoPtr<AudioFloatArray> m_realComponents;
+    nsAutoPtr<AudioFloatArray> m_imagComponents;
+
+    // The lowest frequency (in Hertz) where playback will include all of the
+    // partials.  Playing back lower than this frequency will gradually lose
+    // more high-frequency information.
+    // This frequency is quite low (~10Hz @ // 44.1KHz)
+    float m_lowestFundamentalFrequency;
+
+    float m_rateScale;
+
+    unsigned numberOfRanges() const { return m_numberOfRanges; }
+
+    // Maximum possible number of partials (before culling).
+    unsigned maxNumberOfPartials() const;
+
+    unsigned numberOfPartialsForRange(unsigned rangeIndex) const;
+
+    // Creates table for specified index based on fundamental frequency.
+    void createBandLimitedTables(float fundamentalFrequency, unsigned rangeIndex);
+    unsigned m_maxPartialsInBandLimitedTable;
+    float m_normalizationScale;
+    bool m_disableNormalization;
+    nsTArray<nsAutoPtr<AlignedAudioFloatArray> > m_bandLimitedTables;
+};
+
+} // namespace WebCore
+
+#endif // PeriodicWave_h
diff --git a/dom/media/webaudio/blink/README b/dom/media/webaudio/blink/README
new file mode 100644
index 0000000000..96d209dfc8
--- /dev/null
+++ b/dom/media/webaudio/blink/README
@@ -0,0 +1,24 @@
+This directory contains the code originally borrowed from the Blink Web Audio
+implementation.  We are forking the code here because in many cases the burden
+of adopting Blink specific utilities is too large compared to the prospect of
+importing upstream fixes by just copying newer versions of the code in the
+future.
+
+The process of borrowing code from Blink is as follows:
+
+* Try to borrow utility classes only, and avoid borrowing code which depends
+  too much on the Blink specific utilities.
+* First, import the pristine files from the Blink repository before adding
+  them to the build system, noting the SVN revision of Blink from which the
+  original files were copied in the commit message.
+* In a separate commit, add the imported source files to the build system,
+  and apply the necessary changes to make it build successfully.
+* Use the code in a separate commit.
+* Never add headers as exported headers.  All headers should be included
+  using the following convention: #include "blink/Header.h".
+* Leave the imported code in the WebCore namespace, and import the needed
+  names into the Mozilla code via `using'.
+* Cherry-pick upsteam fixes manually when needed.  In case you fix a problem
+  that is not Mozilla specific locally, try to upstream your changes into
+  Blink.
+* Ping ehsan for any questions.
diff --git a/dom/media/webaudio/blink/Reverb.cpp b/dom/media/webaudio/blink/Reverb.cpp
new file mode 100644
index 0000000000..4fca0822b6
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.cpp
@@ -0,0 +1,243 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "Reverb.h"
+#include "ReverbConvolverStage.h"
+
+#include <math.h>
+#include "ReverbConvolver.h"
+#include "mozilla/FloatingPoint.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+// Empirical gain calibration tested across many impulse responses to ensure perceived volume is same as dry (unprocessed) signal
+const float GainCalibration = -58;
+const float GainCalibrationSampleRate = 44100;
+
+// A minimum power value to when normalizing a silent (or very quiet) impulse response
+const float MinPower = 0.000125f;
+
+static float calculateNormalizationScale(ThreadSharedFloatArrayBufferList* response, size_t aLength, float sampleRate)
+{
+    // Normalize by RMS power
+    size_t numberOfChannels = response->GetChannels();
+
+    float power = 0;
+
+    for (size_t i = 0; i < numberOfChannels; ++i) {
+        float channelPower = AudioBufferSumOfSquares(static_cast<const float*>(response->GetData(i)), aLength);
+        power += channelPower;
+    }
+
+    power = sqrt(power / (numberOfChannels * aLength));
+
+    // Protect against accidental overload
+    if (!IsFinite(power) || IsNaN(power) || power < MinPower)
+        power = MinPower;
+
+    float scale = 1 / power;
+
+    scale *= powf(10, GainCalibration * 0.05f); // calibrate to make perceived volume same as unprocessed
+
+    // Scale depends on sample-rate.
+    if (sampleRate)
+        scale *= GainCalibrationSampleRate / sampleRate;
+
+    // True-stereo compensation
+    if (response->GetChannels() == 4)
+        scale *= 0.5f;
+
+    return scale;
+}
+
+Reverb::Reverb(ThreadSharedFloatArrayBufferList* impulseResponse, size_t impulseResponseBufferLength, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize, float sampleRate)
+{
+    float scale = 1;
+
+    AutoTArray<const float*,4> irChannels;
+    for (size_t i = 0; i < impulseResponse->GetChannels(); ++i) {
+        irChannels.AppendElement(impulseResponse->GetData(i));
+    }
+    AutoTArray<float,1024> tempBuf;
+
+    if (normalize) {
+        scale = calculateNormalizationScale(impulseResponse, impulseResponseBufferLength, sampleRate);
+
+        if (scale) {
+            tempBuf.SetLength(irChannels.Length()*impulseResponseBufferLength);
+            for (uint32_t i = 0; i < irChannels.Length(); ++i) {
+                float* buf = &tempBuf[i*impulseResponseBufferLength];
+                AudioBufferCopyWithScale(irChannels[i], scale, buf,
+                                         impulseResponseBufferLength);
+                irChannels[i] = buf;
+            }
+        }
+    }
+
+    initialize(irChannels, impulseResponseBufferLength,
+               maxFFTSize, numberOfChannels, useBackgroundThreads);
+}
+
+size_t Reverb::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+    amount += m_convolvers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_convolvers.Length(); i++) {
+        if (m_convolvers[i]) {
+            amount += m_convolvers[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    amount += m_tempBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
+    return amount;
+}
+
+
+void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                        size_t impulseResponseBufferLength,
+                        size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+{
+    m_impulseResponseLength = impulseResponseBufferLength;
+
+    // The reverb can handle a mono impulse response and still do stereo processing
+    size_t numResponseChannels = impulseResponseBuffer.Length();
+    m_convolvers.SetCapacity(numberOfChannels);
+
+    int convolverRenderPhase = 0;
+    for (size_t i = 0; i < numResponseChannels; ++i) {
+        const float* channel = impulseResponseBuffer[i];
+        size_t length = impulseResponseBufferLength;
+
+        nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
+        m_convolvers.AppendElement(convolver.forget());
+
+        convolverRenderPhase += WEBAUDIO_BLOCK_SIZE;
+    }
+
+    // For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
+    // It can be bad to allocate memory in a real-time thread.
+    if (numResponseChannels == 4) {
+        m_tempBuffer.AllocateChannels(2);
+        WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
+    }
+}
+
+void Reverb::process(const AudioBlock* sourceBus, AudioBlock* destinationBus)
+{
+    // Do a fairly comprehensive sanity check.
+    // If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
+    bool isSafeToProcess = sourceBus && destinationBus && sourceBus->ChannelCount() > 0 && destinationBus->mChannelData.Length() > 0
+        && WEBAUDIO_BLOCK_SIZE <= MaxFrameSize && WEBAUDIO_BLOCK_SIZE <= size_t(sourceBus->GetDuration()) && WEBAUDIO_BLOCK_SIZE <= size_t(destinationBus->GetDuration());
+
+    MOZ_ASSERT(isSafeToProcess);
+    if (!isSafeToProcess)
+        return;
+
+    // For now only handle mono or stereo output
+    MOZ_ASSERT(destinationBus->ChannelCount() <= 2);
+
+    float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
+    const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
+
+    // Handle input -> output matrixing...
+    size_t numInputChannels = sourceBus->ChannelCount();
+    size_t numOutputChannels = destinationBus->ChannelCount();
+    size_t numReverbChannels = m_convolvers.Length();
+
+    if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
+        // 2 -> 2 -> 2
+        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusR, destinationChannelR);
+    } else  if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
+        // 1 -> 2 -> 2
+        for (int i = 0; i < 2; ++i) {
+            float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
+            m_convolvers[i]->process(sourceBusL, destinationChannel);
+        }
+    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
+        // 1 -> 1 -> 2
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+
+        // simply copy L -> R
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+        bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->GetDuration()) >= WEBAUDIO_BLOCK_SIZE;
+        MOZ_ASSERT(isCopySafe);
+        if (!isCopySafe)
+            return;
+        PodCopy(destinationChannelR, destinationChannelL, WEBAUDIO_BLOCK_SIZE);
+    } else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
+        // 1 -> 1 -> 1
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+    } else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
+        // 2 -> 4 -> 2 ("True" stereo)
+        const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+        // Process left virtual source
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+        // Process right virtual source
+        m_convolvers[2]->process(sourceBusR, tempChannelL);
+        m_convolvers[3]->process(sourceBusR, tempChannelR);
+
+        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
+        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
+    } else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
+        // 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
+        // This is an inefficient use of a four-channel impulse response, but we should handle the case.
+        float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+        float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+        float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+        // Process left virtual source
+        m_convolvers[0]->process(sourceBusL, destinationChannelL);
+        m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+        // Process right virtual source
+        m_convolvers[2]->process(sourceBusL, tempChannelL);
+        m_convolvers[3]->process(sourceBusL, tempChannelR);
+
+        AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
+        AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
+    } else {
+        // Handle gracefully any unexpected / unsupported matrixing
+        // FIXME: add code for 5.1 support...
+        destinationBus->SetNull(destinationBus->GetDuration());
+    }
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/Reverb.h b/dom/media/webaudio/blink/Reverb.h
new file mode 100644
index 0000000000..35e72283dc
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.h
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef Reverb_h
+#define Reverb_h
+
+#include "ReverbConvolver.h"
+#include "nsAutoPtr.h"
+#include "nsTArray.h"
+#include "AudioBlock.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+class ThreadSharedFloatArrayBufferList;
+} // namespace mozilla
+
+namespace WebCore {
+
+// Multi-channel convolution reverb with channel matrixing - one or more ReverbConvolver objects are used internally.
+
+class Reverb {
+public:
+    enum { MaxFrameSize = 256 };
+
+    // renderSliceSize is a rendering hint, so the FFTs can be optimized to not all occur at the same time (very bad when rendering on a real-time thread).
+    Reverb(mozilla::ThreadSharedFloatArrayBufferList* impulseResponseBuffer,
+           size_t impulseResponseBufferLength, size_t maxFFTSize,
+           size_t numberOfChannels, bool useBackgroundThreads, bool normalize,
+           float sampleRate);
+
+    void process(const mozilla::AudioBlock* sourceBus,
+                 mozilla::AudioBlock* destinationBus);
+
+    size_t impulseResponseLength() const { return m_impulseResponseLength; }
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    void initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                    size_t impulseResponseBufferLength, size_t maxFFTSize,
+                    size_t numberOfChannels, bool useBackgroundThreads);
+
+    size_t m_impulseResponseLength;
+
+    nsTArray<nsAutoPtr<ReverbConvolver> > m_convolvers;
+
+    // For "True" stereo processing
+    mozilla::AudioBlock m_tempBuffer;
+};
+
+} // namespace WebCore
+
+#endif // Reverb_h
diff --git a/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp b/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp
new file mode 100644
index 0000000000..4405164b26
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbAccumulationBuffer.h"
+#include "AudioNodeEngine.h"
+#include "mozilla/PodOperations.h"
+#include <algorithm>
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbAccumulationBuffer::ReverbAccumulationBuffer(size_t length)
+    : m_readIndex(0)
+    , m_readTimeFrame(0)
+{
+  m_buffer.SetLength(length);
+  PodZero(m_buffer.Elements(), length);
+}
+
+void ReverbAccumulationBuffer::readAndClear(float* destination, size_t numberOfFrames)
+{
+    size_t bufferLength = m_buffer.Length();
+    bool isCopySafe = m_readIndex <= bufferLength && numberOfFrames <= bufferLength;
+
+    MOZ_ASSERT(isCopySafe);
+    if (!isCopySafe)
+        return;
+
+    size_t framesAvailable = bufferLength - m_readIndex;
+    size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
+    size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
+
+    float* source = m_buffer.Elements();
+    memcpy(destination, source + m_readIndex, sizeof(float) * numberOfFrames1);
+    memset(source + m_readIndex, 0, sizeof(float) * numberOfFrames1);
+
+    // Handle wrap-around if necessary
+    if (numberOfFrames2 > 0) {
+        memcpy(destination + numberOfFrames1, source, sizeof(float) * numberOfFrames2);
+        memset(source, 0, sizeof(float) * numberOfFrames2);
+    }
+
+    m_readIndex = (m_readIndex + numberOfFrames) % bufferLength;
+    m_readTimeFrame += numberOfFrames;
+}
+
+void ReverbAccumulationBuffer::updateReadIndex(int* readIndex, size_t numberOfFrames) const
+{
+    // Update caller's readIndex
+    *readIndex = (*readIndex + numberOfFrames) % m_buffer.Length();
+}
+
+int ReverbAccumulationBuffer::accumulate(const float* source, size_t numberOfFrames, int* readIndex, size_t delayFrames)
+{
+    size_t bufferLength = m_buffer.Length();
+
+    size_t writeIndex = (*readIndex + delayFrames) % bufferLength;
+
+    // Update caller's readIndex
+    *readIndex = (*readIndex + numberOfFrames) % bufferLength;
+
+    size_t framesAvailable = bufferLength - writeIndex;
+    size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
+    size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
+
+    float* destination = m_buffer.Elements();
+
+    bool isSafe = writeIndex <= bufferLength && numberOfFrames1 + writeIndex <= bufferLength && numberOfFrames2 <= bufferLength;
+    MOZ_ASSERT(isSafe);
+    if (!isSafe)
+        return 0;
+
+    AudioBufferAddWithScale(source, 1.0f, destination + writeIndex, numberOfFrames1);
+    if (numberOfFrames2 > 0) {
+        AudioBufferAddWithScale(source + numberOfFrames1, 1.0f, destination, numberOfFrames2);
+    }
+
+    return writeIndex;
+}
+
+void ReverbAccumulationBuffer::reset()
+{
+    PodZero(m_buffer.Elements(), m_buffer.Length());
+    m_readIndex = 0;
+    m_readTimeFrame = 0;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbAccumulationBuffer.h b/dom/media/webaudio/blink/ReverbAccumulationBuffer.h
new file mode 100644
index 0000000000..a37741a2e5
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbAccumulationBuffer.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbAccumulationBuffer_h
+#define ReverbAccumulationBuffer_h
+
+#include "AlignedTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+// ReverbAccumulationBuffer is a circular delay buffer with one client reading from it and multiple clients
+// writing/accumulating to it at different delay offsets from the read position.  The read operation will zero the memory
+// just read from the buffer, so it will be ready for accumulation the next time around.
+class ReverbAccumulationBuffer {
+public:
+    explicit ReverbAccumulationBuffer(size_t length);
+
+    // This will read from, then clear-out numberOfFrames
+    void readAndClear(float* destination, size_t numberOfFrames);
+
+    // Each ReverbConvolverStage will accumulate its output at the appropriate delay from the read position.
+    // We need to pass in and update readIndex here, since each ReverbConvolverStage may be running in
+    // a different thread than the realtime thread calling ReadAndClear() and maintaining m_readIndex
+    // Returns the writeIndex where the accumulation took place
+    int accumulate(const float* source, size_t numberOfFrames, int* readIndex, size_t delayFrames);
+
+    size_t readIndex() const { return m_readIndex; }
+    void updateReadIndex(int* readIndex, size_t numberOfFrames) const;
+
+    size_t readTimeFrame() const { return m_readTimeFrame; }
+
+    void reset();
+
+    size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+    {
+        return m_buffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    }
+
+private:
+    AlignedTArray<float, 16> m_buffer;
+    size_t m_readIndex;
+    size_t m_readTimeFrame; // for debugging (frame on continuous timeline)
+};
+
+} // namespace WebCore
+
+#endif // ReverbAccumulationBuffer_h
diff --git a/dom/media/webaudio/blink/ReverbConvolver.cpp b/dom/media/webaudio/blink/ReverbConvolver.cpp
new file mode 100644
index 0000000000..e739400aee
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolver.cpp
@@ -0,0 +1,265 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbConvolver.h"
+#include "ReverbConvolverStage.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+const int InputBufferSize = 8 * 16384;
+
+// We only process the leading portion of the impulse response in the real-time thread.  We don't exceed this length.
+// It turns out then, that the background thread has about 278msec of scheduling slop.
+// Empirically, this has been found to be a good compromise between giving enough time for scheduling slop,
+// while still minimizing the amount of processing done in the primary (high-priority) thread.
+// This was found to be a good value on Mac OS X, and may work well on other platforms as well, assuming
+// the very rough scheduling latencies are similar on these time-scales.  Of course, this code may need to be
+// tuned for individual platforms if this assumption is found to be incorrect.
+const size_t RealtimeFrameLimit = 8192 + 4096 // ~278msec @ 44.1KHz
+                                  - WEBAUDIO_BLOCK_SIZE;
+// First stage will have size MinFFTSize - successive stages will double in
+// size each time until we hit the maximum size.
+const size_t MinFFTSize = 256;
+// If we are using background threads then don't exceed this FFT size for the
+// stages which run in the real-time thread.  This avoids having only one or
+// two large stages (size 16384 or so) at the end which take a lot of time
+// every several processing slices.  This way we amortize the cost over more
+// processing slices.
+const size_t MaxRealtimeFFTSize = 4096;
+
+ReverbConvolver::ReverbConvolver(const float* impulseResponseData,
+                                 size_t impulseResponseLength,
+                                 size_t maxFFTSize,
+                                 size_t convolverRenderPhase,
+                                 bool useBackgroundThreads)
+    : m_impulseResponseLength(impulseResponseLength)
+    , m_accumulationBuffer(impulseResponseLength + WEBAUDIO_BLOCK_SIZE)
+    , m_inputBuffer(InputBufferSize)
+    , m_backgroundThread("ConvolverWorker")
+    , m_backgroundThreadCondition(&m_backgroundThreadLock)
+    , m_useBackgroundThreads(useBackgroundThreads)
+    , m_wantsToExit(false)
+    , m_moreInputBuffered(false)
+{
+    // For the moment, a good way to know if we have real-time constraint is to check if we're using background threads.
+    // Otherwise, assume we're being run from a command-line tool.
+    bool hasRealtimeConstraint = useBackgroundThreads;
+
+    const float* response = impulseResponseData;
+    size_t totalResponseLength = impulseResponseLength;
+
+    // The total latency is zero because the first FFT stage is small enough
+    // to return output in the first block.
+    size_t reverbTotalLatency = 0;
+
+    size_t stageOffset = 0;
+    size_t stagePhase = 0;
+    size_t fftSize = MinFFTSize;
+    while (stageOffset < totalResponseLength) {
+        size_t stageSize = fftSize / 2;
+
+        // For the last stage, it's possible that stageOffset is such that we're straddling the end
+        // of the impulse response buffer (if we use stageSize), so reduce the last stage's length...
+        if (stageSize + stageOffset > totalResponseLength) {
+            stageSize = totalResponseLength - stageOffset;
+            // Use smallest FFT that is large enough to cover the last stage.
+            fftSize = MinFFTSize;
+            while (stageSize * 2 > fftSize) {
+              fftSize *= 2;
+            }
+        }
+
+        // This "staggers" the time when each FFT happens so they don't all happen at the same time
+        int renderPhase = convolverRenderPhase + stagePhase;
+
+        nsAutoPtr<ReverbConvolverStage> stage
+          (new ReverbConvolverStage(response, totalResponseLength,
+                                    reverbTotalLatency, stageOffset, stageSize,
+                                    fftSize, renderPhase,
+                                    &m_accumulationBuffer));
+
+        bool isBackgroundStage = false;
+
+        if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
+            m_backgroundStages.AppendElement(stage.forget());
+            isBackgroundStage = true;
+        } else
+            m_stages.AppendElement(stage.forget());
+
+        // Figure out next FFT size
+        fftSize *= 2;
+
+        stageOffset += stageSize;
+
+        if (hasRealtimeConstraint && !isBackgroundStage
+            && fftSize > MaxRealtimeFFTSize) {
+            fftSize = MaxRealtimeFFTSize;
+            // Custom phase positions for all but the first of the realtime
+            // stages of largest size.  These spread out the work of the
+            // larger realtime stages.  None of the FFTs of size 1024, 2048 or
+            // 4096 are performed when processing the same block.  The first
+            // MaxRealtimeFFTSize = 4096 stage, at the end of the doubling,
+            // performs its FFT at block 7.  The FFTs of size 2048 are
+            // performed in blocks 3 + 8 * n and size 1024 at 1 + 4 * n.
+            const uint32_t phaseLookup[] = { 14, 0, 10, 4 };
+            stagePhase = WEBAUDIO_BLOCK_SIZE *
+                phaseLookup[m_stages.Length() % ArrayLength(phaseLookup)];
+        } else if (fftSize > maxFFTSize) {
+            fftSize = maxFFTSize;
+            // A prime offset spreads out FFTs in a way that all
+            // available phase positions will be used if there are sufficient
+            // stages.
+            stagePhase += 5 * WEBAUDIO_BLOCK_SIZE;
+        } else if (stageSize > WEBAUDIO_BLOCK_SIZE) {
+            // As the stages are doubling in size, the next FFT will occur
+            // mid-way between FFTs for this stage.
+            stagePhase = stageSize - WEBAUDIO_BLOCK_SIZE;
+        }
+    }
+
+    // Start up background thread
+    // FIXME: would be better to up the thread priority here.  It doesn't need to be real-time, but higher than the default...
+    if (this->useBackgroundThreads() && m_backgroundStages.Length() > 0) {
+        if (!m_backgroundThread.Start()) {
+          NS_WARNING("Cannot start convolver thread.");
+          return;
+        }
+        m_backgroundThread.message_loop()->PostTask(NewNonOwningRunnableMethod(this,
+									       &ReverbConvolver::backgroundThreadEntry));
+    }
+}
+
+ReverbConvolver::~ReverbConvolver()
+{
+    // Wait for background thread to stop
+    if (useBackgroundThreads() && m_backgroundThread.IsRunning()) {
+        m_wantsToExit = true;
+
+        // Wake up thread so it can return
+        {
+            AutoLock locker(m_backgroundThreadLock);
+            m_moreInputBuffered = true;
+            m_backgroundThreadCondition.Signal();
+        }
+
+        m_backgroundThread.Stop();
+    }
+}
+
+size_t ReverbConvolver::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+    amount += m_stages.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_stages.Length(); i++) {
+        if (m_stages[i]) {
+            amount += m_stages[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    amount += m_backgroundStages.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < m_backgroundStages.Length(); i++) {
+        if (m_backgroundStages[i]) {
+            amount += m_backgroundStages[i]->sizeOfIncludingThis(aMallocSizeOf);
+        }
+    }
+
+    // NB: The buffer sizes are static, so even though they might be accessed
+    //     in another thread it's safe to measure them.
+    amount += m_accumulationBuffer.sizeOfExcludingThis(aMallocSizeOf);
+    amount += m_inputBuffer.sizeOfExcludingThis(aMallocSizeOf);
+
+    // Possible future measurements:
+    // - m_backgroundThread
+    // - m_backgroundThreadLock
+    // - m_backgroundThreadCondition
+    return amount;
+}
+
+void ReverbConvolver::backgroundThreadEntry()
+{
+    while (!m_wantsToExit) {
+        // Wait for realtime thread to give us more input
+        m_moreInputBuffered = false;
+        {
+            AutoLock locker(m_backgroundThreadLock);
+            while (!m_moreInputBuffered && !m_wantsToExit)
+                m_backgroundThreadCondition.Wait();
+        }
+
+        // Process all of the stages until their read indices reach the input buffer's write index
+        int writeIndex = m_inputBuffer.writeIndex();
+
+        // Even though it doesn't seem like every stage needs to maintain its own version of readIndex 
+        // we do this in case we want to run in more than one background thread.
+        int readIndex;
+
+        while ((readIndex = m_backgroundStages[0]->inputReadIndex()) != writeIndex) { // FIXME: do better to detect buffer overrun...
+            // Accumulate contributions from each stage
+            for (size_t i = 0; i < m_backgroundStages.Length(); ++i)
+                m_backgroundStages[i]->processInBackground(this);
+        }
+    }
+}
+
+void ReverbConvolver::process(const float* sourceChannelData,
+                              float* destinationChannelData)
+{
+    const float* source = sourceChannelData;
+    float* destination = destinationChannelData;
+    bool isDataSafe = source && destination;
+    MOZ_ASSERT(isDataSafe);
+    if (!isDataSafe)
+        return;
+
+    // Feed input buffer (read by all threads)
+    m_inputBuffer.write(source, WEBAUDIO_BLOCK_SIZE);
+
+    // Accumulate contributions from each stage
+    for (size_t i = 0; i < m_stages.Length(); ++i)
+        m_stages[i]->process(source);
+
+    // Finally read from accumulation buffer
+    m_accumulationBuffer.readAndClear(destination, WEBAUDIO_BLOCK_SIZE);
+
+    // Now that we've buffered more input, wake up our background thread.
+
+    // Not using a MutexLocker looks strange, but we use a tryLock() instead because this is run on the real-time
+    // thread where it is a disaster for the lock to be contended (causes audio glitching).  It's OK if we fail to
+    // signal from time to time, since we'll get to it the next time we're called.  We're called repeatedly
+    // and frequently (around every 3ms).  The background thread is processing well into the future and has a considerable amount of 
+    // leeway here...
+    if (m_backgroundThreadLock.Try()) {
+        m_moreInputBuffered = true;
+        m_backgroundThreadCondition.Signal();
+        m_backgroundThreadLock.Release();
+    }
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbConvolver.h b/dom/media/webaudio/blink/ReverbConvolver.h
new file mode 100644
index 0000000000..b7eea45b86
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolver.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbConvolver_h
+#define ReverbConvolver_h
+
+#include "ReverbAccumulationBuffer.h"
+#include "ReverbInputBuffer.h"
+#include "nsAutoPtr.h"
+#include "mozilla/MemoryReporting.h"
+#ifdef LOG
+#undef LOG
+#endif
+#include "base/condition_variable.h"
+#include "base/lock.h"
+#include "base/thread.h"
+
+namespace WebCore {
+
+class ReverbConvolverStage;
+
+class ReverbConvolver {
+public:
+    // maxFFTSize can be adjusted (from say 2048 to 32768) depending on how much precision is necessary.
+    // For certain tweaky de-convolving applications the phase errors add up quickly and lead to non-sensical results with
+    // larger FFT sizes and single-precision floats.  In these cases 2048 is a good size.
+    // If not doing multi-threaded convolution, then should not go > 8192.
+    ReverbConvolver(const float* impulseResponseData,
+                    size_t impulseResponseLength, size_t maxFFTSize,
+                    size_t convolverRenderPhase, bool useBackgroundThreads);
+    ~ReverbConvolver();
+
+    void process(const float* sourceChannelData,
+                 float* destinationChannelData);
+
+    size_t impulseResponseLength() const { return m_impulseResponseLength; }
+
+    ReverbInputBuffer* inputBuffer() { return &m_inputBuffer; }
+
+    bool useBackgroundThreads() const { return m_useBackgroundThreads; }
+    void backgroundThreadEntry();
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+private:
+    nsTArray<nsAutoPtr<ReverbConvolverStage> > m_stages;
+    nsTArray<nsAutoPtr<ReverbConvolverStage> > m_backgroundStages;
+    size_t m_impulseResponseLength;
+
+    ReverbAccumulationBuffer m_accumulationBuffer;
+
+    // One or more background threads read from this input buffer which is fed from the realtime thread.
+    ReverbInputBuffer m_inputBuffer;
+
+    // Background thread and synchronization
+    base::Thread m_backgroundThread;
+    Lock m_backgroundThreadLock;
+    ConditionVariable m_backgroundThreadCondition;
+    bool m_useBackgroundThreads;
+    bool m_wantsToExit;
+    bool m_moreInputBuffered;
+};
+
+} // namespace WebCore
+
+#endif // ReverbConvolver_h
diff --git a/dom/media/webaudio/blink/ReverbConvolverStage.cpp b/dom/media/webaudio/blink/ReverbConvolverStage.cpp
new file mode 100644
index 0000000000..055098e887
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolverStage.cpp
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbConvolverStage.h"
+
+#include "ReverbAccumulationBuffer.h"
+#include "ReverbConvolver.h"
+#include "ReverbInputBuffer.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t,
+                                           size_t reverbTotalLatency,
+                                           size_t stageOffset,
+                                           size_t stageLength,
+                                           size_t fftSize, size_t renderPhase,
+                                           ReverbAccumulationBuffer* accumulationBuffer)
+    : m_accumulationBuffer(accumulationBuffer)
+    , m_accumulationReadIndex(0)
+    , m_inputReadIndex(0)
+{
+    MOZ_ASSERT(impulseResponse);
+    MOZ_ASSERT(accumulationBuffer);
+
+    m_fftKernel = new FFTBlock(fftSize);
+    m_fftKernel->PadAndMakeScaledDFT(impulseResponse + stageOffset, stageLength);
+    m_fftConvolver = new FFTConvolver(fftSize, renderPhase);
+
+    // The convolution stage at offset stageOffset needs to have a corresponding delay to cancel out the offset.
+    size_t totalDelay = stageOffset + reverbTotalLatency;
+
+    // But, the FFT convolution itself incurs latency, so subtract this out...
+    size_t fftLatency = m_fftConvolver->latencyFrames();
+    MOZ_ASSERT(totalDelay >= fftLatency);
+    totalDelay -= fftLatency;
+
+    m_postDelayLength = totalDelay;
+}
+
+size_t ReverbConvolverStage::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+{
+    size_t amount = aMallocSizeOf(this);
+
+    if (m_fftKernel) {
+        amount += m_fftKernel->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    if (m_fftConvolver) {
+        amount += m_fftConvolver->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+}
+
+void ReverbConvolverStage::processInBackground(ReverbConvolver* convolver)
+{
+    ReverbInputBuffer* inputBuffer = convolver->inputBuffer();
+    float* source = inputBuffer->directReadFrom(&m_inputReadIndex,
+                                                WEBAUDIO_BLOCK_SIZE);
+    process(source);
+}
+
+void ReverbConvolverStage::process(const float* source)
+{
+    MOZ_ASSERT(source);
+    if (!source)
+        return;
+    
+    // Now, run the convolution (into the delay buffer).
+    // An expensive FFT will happen every fftSize / 2 frames.
+    const float* output = m_fftConvolver->process(m_fftKernel, source);
+
+    // Now accumulate into reverb's accumulation buffer.
+    m_accumulationBuffer->accumulate(output, WEBAUDIO_BLOCK_SIZE,
+                                     &m_accumulationReadIndex,
+                                     m_postDelayLength);
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbConvolverStage.h b/dom/media/webaudio/blink/ReverbConvolverStage.h
new file mode 100644
index 0000000000..0ebc33f3a8
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolverStage.h
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbConvolverStage_h
+#define ReverbConvolverStage_h
+
+#include "FFTConvolver.h"
+
+#include "nsAutoPtr.h"
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+using mozilla::FFTBlock;
+
+class ReverbAccumulationBuffer;
+class ReverbConvolver;
+
+// A ReverbConvolverStage represents the convolution associated with a sub-section of a large impulse response.
+// It incorporates a delay line to account for the offset of the sub-section within the larger impulse response.
+class ReverbConvolverStage {
+public:
+    // renderPhase is useful to know so that we can manipulate the pre versus post delay so that stages will perform
+    // their heavy work (FFT processing) on different slices to balance the load in a real-time thread.
+    ReverbConvolverStage(const float* impulseResponse, size_t responseLength, size_t reverbTotalLatency, size_t stageOffset, size_t stageLength, size_t fftSize, size_t renderPhase, ReverbAccumulationBuffer*);
+
+    // |source| must point to an array of WEBAUDIO_BLOCK_SIZE elements.
+    void process(const float* source);
+
+    void processInBackground(ReverbConvolver* convolver);
+
+    // Useful for background processing
+    int inputReadIndex() const { return m_inputReadIndex; }
+
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+private:
+    nsAutoPtr<FFTBlock> m_fftKernel;
+    nsAutoPtr<FFTConvolver> m_fftConvolver;
+
+    ReverbAccumulationBuffer* m_accumulationBuffer;
+    int m_accumulationReadIndex;
+    int m_inputReadIndex;
+
+    size_t m_postDelayLength;
+
+    nsTArray<float> m_temporaryBuffer;
+};
+
+} // namespace WebCore
+
+#endif // ReverbConvolverStage_h
diff --git a/dom/media/webaudio/blink/ReverbInputBuffer.cpp b/dom/media/webaudio/blink/ReverbInputBuffer.cpp
new file mode 100644
index 0000000000..8221f8151d
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbInputBuffer.cpp
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbInputBuffer.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbInputBuffer::ReverbInputBuffer(size_t length)
+    : m_writeIndex(0)
+{
+  m_buffer.SetLength(length);
+  PodZero(m_buffer.Elements(), length);
+}
+
+void ReverbInputBuffer::write(const float* sourceP, size_t numberOfFrames)
+{
+    size_t bufferLength = m_buffer.Length();
+    bool isCopySafe = m_writeIndex + numberOfFrames <= bufferLength;
+    MOZ_ASSERT(isCopySafe);
+    if (!isCopySafe)
+        return;
+
+    memcpy(m_buffer.Elements() + m_writeIndex, sourceP, sizeof(float) * numberOfFrames);
+
+    m_writeIndex += numberOfFrames;
+    MOZ_ASSERT(m_writeIndex <= bufferLength);
+
+    if (m_writeIndex >= bufferLength)
+        m_writeIndex = 0;
+}
+
+float* ReverbInputBuffer::directReadFrom(int* readIndex, size_t numberOfFrames)
+{
+    size_t bufferLength = m_buffer.Length();
+    bool isPointerGood = readIndex && *readIndex >= 0 && *readIndex + numberOfFrames <= bufferLength;
+    MOZ_ASSERT(isPointerGood);
+    if (!isPointerGood) {
+        // Should never happen in practice but return pointer to start of buffer (avoid crash)
+        if (readIndex)
+            *readIndex = 0;
+        return m_buffer.Elements();
+    }
+
+    float* sourceP = m_buffer.Elements();
+    float* p = sourceP + *readIndex;
+
+    // Update readIndex
+    *readIndex = (*readIndex + numberOfFrames) % bufferLength;
+
+    return p;
+}
+
+void ReverbInputBuffer::reset()
+{
+    PodZero(m_buffer.Elements(), m_buffer.Length());
+    m_writeIndex = 0;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbInputBuffer.h b/dom/media/webaudio/blink/ReverbInputBuffer.h
new file mode 100644
index 0000000000..906021c0d1
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbInputBuffer.h
@@ -0,0 +1,71 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbInputBuffer_h
+#define ReverbInputBuffer_h
+
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+// ReverbInputBuffer is used to buffer input samples for deferred processing by the background threads.
+class ReverbInputBuffer {
+public:
+    explicit ReverbInputBuffer(size_t length);
+
+    // The realtime audio thread keeps writing samples here.
+    // The assumption is that the buffer's length is evenly divisible by numberOfFrames (for nearly all cases this will be fine).
+    // FIXME: remove numberOfFrames restriction...
+    void write(const float* sourceP, size_t numberOfFrames);
+
+    // Background threads can call this to check if there's anything to read...
+    size_t writeIndex() const { return m_writeIndex; }
+
+    // The individual background threads read here (and hope that they can keep up with the buffer writing).
+    // readIndex is updated with the next readIndex to read from...
+    // The assumption is that the buffer's length is evenly divisible by numberOfFrames.
+    // FIXME: remove numberOfFrames restriction...
+    float* directReadFrom(int* readIndex, size_t numberOfFrames);
+
+    void reset();
+
+    size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const
+    {
+        return m_buffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    }
+
+
+private:
+    nsTArray<float> m_buffer;
+    size_t m_writeIndex;
+};
+
+} // namespace WebCore
+
+#endif // ReverbInputBuffer_h
diff --git a/dom/media/webaudio/blink/ZeroPole.cpp b/dom/media/webaudio/blink/ZeroPole.cpp
new file mode 100644
index 0000000000..ac0b15c7ad
--- /dev/null
+++ b/dom/media/webaudio/blink/ZeroPole.cpp
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ZeroPole.h"
+
+#include <cmath>
+#include <float.h>
+
+namespace WebCore {
+
+void ZeroPole::process(const float *source, float *destination, int framesToProcess)
+{
+    float zero = m_zero;
+    float pole = m_pole;
+
+    // Gain compensation to make 0dB @ 0Hz
+    const float k1 = 1 / (1 - zero);
+    const float k2 = 1 - pole;
+    
+    // Member variables to locals.
+    float lastX = m_lastX;
+    float lastY = m_lastY;
+
+    for (int i = 0; i < framesToProcess; ++i) {
+        float input = source[i];
+
+        // Zero
+        float output1 = k1 * (input - zero * lastX);
+        lastX = input;
+
+        // Pole
+        float output2 = k2 * output1 + pole * lastY;
+        lastY = output2;
+
+        destination[i] = output2;
+    }
+    
+    // Locals to member variables. Flush denormals here so we don't
+    // slow down the inner loop above.
+    if (lastX == 0.0f && lastY != 0.0f && fabsf(lastY) < FLT_MIN) {
+        // Flush future values to zero (until there is new input).
+        lastY = 0.0;
+        // Flush calculated values.
+        for (int i = framesToProcess; i-- && fabsf(destination[i]) < FLT_MIN; ) {
+            destination[i] = 0.0f;
+        }
+    }
+
+    m_lastX = lastX;
+    m_lastY = lastY;
+}
+
+} // namespace WebCore
diff --git a/dom/media/webaudio/blink/ZeroPole.h b/dom/media/webaudio/blink/ZeroPole.h
new file mode 100644
index 0000000000..7381bde3d2
--- /dev/null
+++ b/dom/media/webaudio/blink/ZeroPole.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ZeroPole_h
+#define ZeroPole_h
+
+namespace WebCore {
+
+// ZeroPole is a simple filter with one zero and one pole.
+
+class ZeroPole {
+public:
+    ZeroPole()
+        : m_zero(0)
+        , m_pole(0)
+        , m_lastX(0)
+        , m_lastY(0)
+    {
+    }
+
+    void process(const float *source, float *destination, int framesToProcess);
+
+    // Reset filter state.
+    void reset() { m_lastX = 0; m_lastY = 0; }
+    
+    void setZero(float zero) { m_zero = zero; }
+    void setPole(float pole) { m_pole = pole; }
+    
+    float zero() const { return m_zero; }
+    float pole() const { return m_pole; }
+
+private:
+    float m_zero;
+    float m_pole;
+    float m_lastX;
+    float m_lastY;
+};
+
+} // namespace WebCore
+
+#endif // ZeroPole_h
diff --git a/dom/media/webaudio/blink/moz.build b/dom/media/webaudio/blink/moz.build
new file mode 100644
index 0000000000..385614de7b
--- /dev/null
+++ b/dom/media/webaudio/blink/moz.build
@@ -0,0 +1,39 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+UNIFIED_SOURCES += [
+    'Biquad.cpp',
+    'DynamicsCompressor.cpp',
+    'DynamicsCompressorKernel.cpp',
+    'FFTConvolver.cpp',
+    'HRTFDatabase.cpp',
+    'HRTFDatabaseLoader.cpp',
+    'HRTFElevation.cpp',
+    'HRTFKernel.cpp',
+    'HRTFPanner.cpp',
+    'IIRFilter.cpp',
+    'PeriodicWave.cpp',
+    'Reverb.cpp',
+    'ReverbAccumulationBuffer.cpp',
+    'ReverbConvolver.cpp',
+    'ReverbConvolverStage.cpp',
+    'ReverbInputBuffer.cpp',
+    'ZeroPole.cpp',
+]
+
+# Are we targeting x86 or x64?  If so, build SSE2 files.
+if CONFIG['INTEL_ARCHITECTURE']:
+    DEFINES['USE_SSE2'] = True
+
+include('/ipc/chromium/chromium-config.mozbuild')
+
+FINAL_LIBRARY = 'xul'
+LOCAL_INCLUDES += [
+    '/dom/media/webaudio',
+]
+
+if CONFIG['GNU_CXX']:
+    CXXFLAGS += ['-Wno-shadow']
diff --git a/dom/media/webaudio/gtest/TestAudioEventTimeline.cpp b/dom/media/webaudio/gtest/TestAudioEventTimeline.cpp
new file mode 100644
index 0000000000..cc731d3e2b
--- /dev/null
+++ b/dom/media/webaudio/gtest/TestAudioEventTimeline.cpp
@@ -0,0 +1,450 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioEventTimeline.h"
+#include <sstream>
+#include <limits>
+#include "gtest/gtest.h"
+
+// Mock the MediaStream class
+namespace mozilla {
+class MediaStream
+{
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaStream)
+private:
+  ~MediaStream() {
+  };
+};
+}
+
+using namespace mozilla;
+using namespace mozilla::dom;
+using std::numeric_limits;
+
+// Some simple testing primitives
+void ok(bool val, const char* msg)
+{
+  if (!val) {
+    fprintf(stderr, "failure: %s", msg);
+  }
+  ASSERT_TRUE(val);
+}
+
+namespace std {
+
+template <class T>
+basic_ostream<T, char_traits<T> >&
+operator<<(basic_ostream<T, char_traits<T> >& os, nsresult rv)
+{
+  os << static_cast<uint32_t>(rv);
+  return os;
+}
+
+} // namespace std
+
+template <class T, class U>
+void is(const T& a, const U& b, const char* msg)
+{
+  std::stringstream ss;
+  ss << msg << ", Got: " << a << ", expected: " << b << std::endl;
+  ok(a == b, ss.str().c_str());
+}
+
+template <>
+void is(const float& a, const float& b, const char* msg)
+{
+  // stupidly high, since we mostly care about the correctness of the algorithm
+  const float kEpsilon = 0.00001f;
+
+  std::stringstream ss;
+  ss << msg << ", Got: " << a << ", expected: " << b << std::endl;
+  ok(fabsf(a - b) < kEpsilon, ss.str().c_str());
+}
+
+class ErrorResultMock
+{
+public:
+  ErrorResultMock()
+    : mRv(NS_OK)
+  {
+  }
+  void Throw(nsresult aRv)
+  {
+    mRv = aRv;
+  }
+
+  operator nsresult() const
+  {
+    return mRv;
+  }
+
+  ErrorResultMock& operator=(nsresult aRv)
+  {
+    mRv = aRv;
+    return *this;
+  }
+
+private:
+  nsresult mRv;
+};
+
+typedef AudioEventTimeline Timeline;
+
+TEST(AudioEventTimeline, SpecExample)
+{
+  // First, run the basic tests
+  Timeline timeline(10.0f);
+  is(timeline.Value(), 10.0f, "Correct default value returned");
+
+  ErrorResultMock rv;
+
+  uint32_t curveLength = 44100;
+  float* curve = new float[curveLength];
+  for (uint32_t i = 0; i < curveLength; ++i) {
+    curve[i] = sin(M_PI * i / float(curveLength));
+  }
+
+  // This test is copied from the example in the Web Audio spec
+  const double t0 = 0.0,
+               t1 = 0.1,
+               t2 = 0.2,
+               t3 = 0.3,
+               t4 = 0.4,
+               t5 = 0.6,
+               t6 = 0.7,
+               t7 = 1.0;
+  timeline.SetValueAtTime(0.2f, t0, rv);
+  is(rv, NS_OK, "SetValueAtTime succeeded");
+  timeline.SetValueAtTime(0.3f, t1, rv);
+  is(rv, NS_OK, "SetValueAtTime succeeded");
+  timeline.SetValueAtTime(0.4f, t2, rv);
+  is(rv, NS_OK, "SetValueAtTime succeeded");
+  timeline.LinearRampToValueAtTime(1.0f, t3, rv);
+  is(rv, NS_OK, "LinearRampToValueAtTime succeeded");
+  timeline.LinearRampToValueAtTime(0.15f, t4, rv);
+  is(rv, NS_OK, "LinearRampToValueAtTime succeeded");
+  timeline.ExponentialRampToValueAtTime(0.75f, t5, rv);
+  is(rv, NS_OK, "ExponentialRampToValueAtTime succeeded");
+  timeline.ExponentialRampToValueAtTime(0.05f, t6, rv);
+  is(rv, NS_OK, "ExponentialRampToValueAtTime succeeded");
+  timeline.SetValueCurveAtTime(curve, curveLength, t6, t7 - t6, rv);
+  is(rv, NS_OK, "SetValueCurveAtTime succeeded");
+
+  is(timeline.GetValueAtTime(0.0), 0.2f, "Correct value");
+  is(timeline.GetValueAtTime(0.05), 0.2f, "Correct value");
+  is(timeline.GetValueAtTime(0.1), 0.3f, "Correct value");
+  is(timeline.GetValueAtTime(0.15), 0.3f, "Correct value");
+  is(timeline.GetValueAtTime(0.2), 0.4f, "Correct value");
+  is(timeline.GetValueAtTime(0.25), (0.4f + 1.0f) / 2, "Correct value");
+  is(timeline.GetValueAtTime(0.3), 1.0f, "Correct value");
+  is(timeline.GetValueAtTime(0.35), (1.0f + 0.15f) / 2, "Correct value");
+  is(timeline.GetValueAtTime(0.4), 0.15f, "Correct value");
+  is(timeline.GetValueAtTime(0.45), (0.15f * powf(0.75f / 0.15f, 0.05f / 0.2f)), "Correct value");
+  is(timeline.GetValueAtTime(0.5), (0.15f * powf(0.75f / 0.15f, 0.5f)), "Correct value");
+  is(timeline.GetValueAtTime(0.55), (0.15f * powf(0.75f / 0.15f, 0.15f / 0.2f)), "Correct value");
+  is(timeline.GetValueAtTime(0.6), 0.75f, "Correct value");
+  is(timeline.GetValueAtTime(0.65), (0.75f * powf(0.05f / 0.75f, 0.5f)), "Correct value");
+  is(timeline.GetValueAtTime(0.7), 0.0f, "Correct value");
+  is(timeline.GetValueAtTime(0.85), 1.0f, "Correct value");
+  is(timeline.GetValueAtTime(1.0), curve[curveLength - 1], "Correct value");
+
+  delete[] curve;
+}
+
+TEST(AudioEventTimeline, InvalidEvents)
+{
+  static_assert(numeric_limits<float>::has_quiet_NaN, "Platform must have a quiet NaN");
+  const float NaN = numeric_limits<float>::quiet_NaN();
+  const float Infinity = numeric_limits<float>::infinity();
+  Timeline timeline(10.0f);
+
+  float curve[] = { -1.0f, 0.0f, 1.0f };
+  float badCurve1[] = { -1.0f, NaN, 1.0f };
+  float badCurve2[] = { -1.0f, Infinity, 1.0f };
+  float badCurve3[] = { -1.0f, -Infinity, 1.0f };
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(NaN, 0.1, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueAtTime(Infinity, 0.1, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueAtTime(-Infinity, 0.1, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.LinearRampToValueAtTime(NaN, 0.2, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.LinearRampToValueAtTime(Infinity, 0.2, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.LinearRampToValueAtTime(-Infinity, 0.2, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.ExponentialRampToValueAtTime(NaN, 0.3, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.ExponentialRampToValueAtTime(Infinity, 0.3, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.ExponentialRampToValueAtTime(-Infinity, 0.4, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.ExponentialRampToValueAtTime(0, 0.5, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(NaN, 0.4, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(Infinity, 0.4, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(-Infinity, 0.4, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(0.4f, NaN, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(0.4f, Infinity, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetTargetAtTime(0.4f, -Infinity, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(nullptr, 0, 1.0, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(badCurve1, ArrayLength(badCurve1), 1.0, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(badCurve2, ArrayLength(badCurve2), 1.0, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(badCurve3, ArrayLength(badCurve3), 1.0, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), NaN, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), Infinity, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), -Infinity, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), 1.0, NaN, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), 1.0, Infinity, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), 1.0, -Infinity, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+}
+
+TEST(AudioEventTimeline, EventReplacement)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  is(timeline.GetEventCount(), 0u, "No events yet");
+  timeline.SetValueAtTime(10.0f, 0.1, rv);
+  is(timeline.GetEventCount(), 1u, "One event scheduled now");
+  timeline.SetValueAtTime(20.0f, 0.1, rv);
+  is(rv, NS_OK, "Event scheduling should be successful");
+  is(timeline.GetEventCount(), 1u, "Event should be replaced");
+  is(timeline.GetValueAtTime(0.1), 20.0f, "The first event should be overwritten");
+  timeline.LinearRampToValueAtTime(30.0f, 0.1, rv);
+  is(rv, NS_OK, "Event scheduling should be successful");
+  is(timeline.GetEventCount(), 2u, "Different event type should be appended");
+  is(timeline.GetValueAtTime(0.1), 30.0f, "The first event should be overwritten");
+}
+
+TEST(AudioEventTimeline, EventRemoval)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(10.0f, 0.1, rv);
+  timeline.SetValueAtTime(15.0f, 0.15, rv);
+  timeline.SetValueAtTime(20.0f, 0.2, rv);
+  timeline.LinearRampToValueAtTime(30.0f, 0.3, rv);
+  is(timeline.GetEventCount(), 4u, "Should have three events initially");
+  timeline.CancelScheduledValues(0.4);
+  is(timeline.GetEventCount(), 4u, "Trying to delete past the end of the array should have no effect");
+  timeline.CancelScheduledValues(0.3);
+  is(timeline.GetEventCount(), 3u, "Should successfully delete one event");
+  timeline.CancelScheduledValues(0.12);
+  is(timeline.GetEventCount(), 1u, "Should successfully delete two events");
+  timeline.CancelAllEvents();
+  ok(timeline.HasSimpleValue(), "No event should remain scheduled");
+}
+
+TEST(AudioEventTimeline, BeforeFirstEventSetValue)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(0.5), 10.0f, "Retrun the default value before the first event");
+}
+
+TEST(AudioEventTimeline, BeforeFirstEventSetTarget)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetTargetAtTime(20.0f, 1.0, 5.0, rv);
+  is(timeline.GetValueAtTime(0.5), 10.0f, "Retrun the default value before the first event");
+}
+
+TEST(AudioEventTimeline, BeforeFirstEventLinearRamp)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.LinearRampToValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(0.5), 10.0f, "Retrun the default value before the first event");
+}
+
+TEST(AudioEventTimeline, BeforeFirstEventExponentialRamp)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.ExponentialRampToValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(0.5), 10.0f, "Retrun the default value before the first event");
+}
+
+TEST(AudioEventTimeline, AfterLastValueEvent)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(1.5), 20.0f, "Return the last value after the last SetValue event");
+}
+
+TEST(AudioEventTimeline, AfterLastTargetValueEvent)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetTargetAtTime(20.0f, 1.0, 5.0, rv);
+  is(timeline.GetValueAtTime(10.), (20.f + (10.f - 20.f) * expf(-9.0f / 5.0f)), "Return the value after the last SetTarget event based on the curve");
+}
+
+TEST(AudioEventTimeline, AfterLastTargetValueEventWithValueSet)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValue(50.f);
+  timeline.SetTargetAtTime(20.0f, 1.0, 5.0, rv);
+
+  // When using SetTargetValueAtTime, Timeline become stateful: the value for
+  // time t may depend on the time t-1, so we can't just query the value at a
+  // time and get the right value. We have to call GetValueAtTime for the
+  // previous times.
+  for (double i = 0.0; i < 9.99; i+=0.01) {
+    timeline.GetValueAtTime(i);
+  }
+
+  is(timeline.GetValueAtTime(10.), (20.f + (50.f - 20.f) * expf(-9.0f / 5.0f)), "Return the value after SetValue and the last SetTarget event based on the curve");
+}
+
+TEST(AudioEventTimeline, Value)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  is(timeline.Value(), 10.0f, "value should initially match the default value");
+  timeline.SetValue(20.0f);
+  is(timeline.Value(), 20.0f, "Should be able to set the value");
+  timeline.SetValueAtTime(20.0f, 1.0, rv);
+  // TODO: The following check needs to change when we compute the value based on the current time of the context
+  is(timeline.Value(), 20.0f, "TODO...");
+  timeline.SetValue(30.0f);
+  is(timeline.Value(), 20.0f, "Should not be able to set the value");
+}
+
+TEST(AudioEventTimeline, LinearRampAtZero)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.LinearRampToValueAtTime(20.0f, 0.0, rv);
+  is(timeline.GetValueAtTime(0.0), 20.0f, "Should get the correct value when t0 == t1 == 0");
+}
+
+TEST(AudioEventTimeline, ExponentialRampAtZero)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.ExponentialRampToValueAtTime(20.0f, 0.0, rv);
+  is(timeline.GetValueAtTime(0.0), 20.0f, "Should get the correct value when t0 == t1 == 0");
+}
+
+TEST(AudioEventTimeline, LinearRampAtSameTime)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(5.0f, 1.0, rv);
+  timeline.LinearRampToValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(1.0), 20.0f, "Should get the correct value when t0 == t1");
+}
+
+TEST(AudioEventTimeline, ExponentialRampAtSameTime)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetValueAtTime(5.0f, 1.0, rv);
+  timeline.ExponentialRampToValueAtTime(20.0f, 1.0, rv);
+  is(timeline.GetValueAtTime(1.0), 20.0f, "Should get the correct value when t0 == t1");
+}
+
+TEST(AudioEventTimeline, SetTargetZeroTimeConstant)
+{
+  Timeline timeline(10.0f);
+
+  ErrorResultMock rv;
+
+  timeline.SetTargetAtTime(20.0f, 1.0, 0.0, rv);
+  is(timeline.GetValueAtTime(1.0), 20.0f, "Should get the correct value when t0 == t1");
+}
+
+TEST(AudioEventTimeline, ExponentialInvalidPreviousZeroValue)
+{
+  Timeline timeline(0.f);
+
+  ErrorResultMock rv;
+
+  timeline.ExponentialRampToValueAtTime(1.f, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.SetValue(1.f);
+  rv = NS_OK;
+  timeline.ExponentialRampToValueAtTime(1.f, 1.0, rv);
+  is(rv, NS_OK, "Should succeed this time");
+  timeline.CancelScheduledValues(0.0);
+  is(timeline.GetEventCount(), 0u, "Should have no events scheduled");
+  rv = NS_OK;
+  timeline.SetValueAtTime(0.f, 0.5, rv);
+  is(rv, NS_OK, "Should succeed");
+  timeline.ExponentialRampToValueAtTime(1.f, 1.0, rv);
+  is(rv, NS_ERROR_DOM_SYNTAX_ERR, "Correct error code returned");
+  timeline.CancelScheduledValues(0.0);
+  is(timeline.GetEventCount(), 0u, "Should have no events scheduled");
+  rv = NS_OK;
+  timeline.ExponentialRampToValueAtTime(1.f, 1.0, rv);
+  is(rv, NS_OK, "Should succeed this time");
+}
+
+TEST(AudioEventTimeline, SettingValueCurveTwice)
+{
+  Timeline timeline(0.f);
+  float curve[] = { -1.0f, 0.0f, 1.0f };
+
+  ErrorResultMock rv;
+
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), 0.0f, 0.3f, rv);
+  timeline.SetValueCurveAtTime(curve, ArrayLength(curve), 0.0f, 0.3f, rv);
+  is(rv, NS_OK, "SetValueCurveAtTime succeeded");
+}
+
diff --git a/dom/media/webaudio/gtest/moz.build b/dom/media/webaudio/gtest/moz.build
new file mode 100644
index 0000000000..2cc13b038d
--- /dev/null
+++ b/dom/media/webaudio/gtest/moz.build
@@ -0,0 +1,15 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+UNIFIED_SOURCES += [
+    'TestAudioEventTimeline.cpp',
+]
+
+LOCAL_INCLUDES += [
+    '..',
+]
+
+FINAL_LIBRARY = 'xul-gtest'
diff --git a/dom/media/webaudio/moz.build b/dom/media/webaudio/moz.build
new file mode 100644
index 0000000000..d1a9f56809
--- /dev/null
+++ b/dom/media/webaudio/moz.build
@@ -0,0 +1,142 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+with Files('*'):
+    BUG_COMPONENT = ('Core', 'Web Audio')
+
+DIRS += ['blink']
+
+TEST_DIRS += ['gtest']
+
+MOCHITEST_MANIFESTS += [
+    'test/blink/mochitest.ini',
+    'test/mochitest.ini',
+]
+
+BROWSER_CHROME_MANIFESTS += [
+    'test/browser.ini',
+]
+
+TEST_HARNESS_FILES.testing.mochitest.tests.dom.media.webaudio.test.blink += [
+    'test/blink/audio-testing.js',
+    'test/blink/convolution-testing.js',
+    'test/blink/panner-model-testing.js',
+]
+
+EXPORTS += [
+    'AlignedTArray.h',
+    'AudioBlock.h',
+    'AudioEventTimeline.h',
+    'AudioNodeEngine.h',
+    'AudioNodeExternalInputStream.h',
+    'AudioNodeStream.h',
+    'AudioParamTimeline.h',
+    'MediaBufferDecoder.h',
+    'ThreeDPoint.h',
+    'WebAudioUtils.h',
+]
+
+EXPORTS.mozilla += [
+    'FFTBlock.h',
+    'MediaStreamAudioDestinationNode.h',
+]
+
+EXPORTS.mozilla.dom += [
+    'AnalyserNode.h',
+    'AudioBuffer.h',
+    'AudioBufferSourceNode.h',
+    'AudioContext.h',
+    'AudioDestinationNode.h',
+    'AudioListener.h',
+    'AudioNode.h',
+    'AudioParam.h',
+    'AudioProcessingEvent.h',
+    'BiquadFilterNode.h',
+    'ChannelMergerNode.h',
+    'ChannelSplitterNode.h',
+    'ConstantSourceNode.h',
+    'ConvolverNode.h',
+    'DelayNode.h',
+    'DynamicsCompressorNode.h',
+    'GainNode.h',
+    'IIRFilterNode.h',
+    'MediaElementAudioSourceNode.h',
+    'MediaStreamAudioDestinationNode.h',
+    'MediaStreamAudioSourceNode.h',
+    'OfflineAudioCompletionEvent.h',
+    'OscillatorNode.h',
+    'PannerNode.h',
+    'PeriodicWave.h',
+    'ScriptProcessorNode.h',
+    'StereoPannerNode.h',
+    'WaveShaperNode.h',
+]
+
+UNIFIED_SOURCES += [
+    'AnalyserNode.cpp',
+    'AudioBlock.cpp',
+    'AudioBuffer.cpp',
+    'AudioBufferSourceNode.cpp',
+    'AudioContext.cpp',
+    'AudioDestinationNode.cpp',
+    'AudioEventTimeline.cpp',
+    'AudioListener.cpp',
+    'AudioNode.cpp',
+    'AudioNodeEngine.cpp',
+    'AudioNodeExternalInputStream.cpp',
+    'AudioNodeStream.cpp',
+    'AudioParam.cpp',
+    'AudioProcessingEvent.cpp',
+    'BiquadFilterNode.cpp',
+    'BufferDecoder.cpp',
+    'ChannelMergerNode.cpp',
+    'ChannelSplitterNode.cpp',
+    'ConstantSourceNode.cpp',
+    'ConvolverNode.cpp',
+    'DelayBuffer.cpp',
+    'DelayNode.cpp',
+    'DynamicsCompressorNode.cpp',
+    'FFTBlock.cpp',
+    'GainNode.cpp',
+    'IIRFilterNode.cpp',
+    'MediaBufferDecoder.cpp',
+    'MediaElementAudioSourceNode.cpp',
+    'MediaStreamAudioDestinationNode.cpp',
+    'MediaStreamAudioSourceNode.cpp',
+    'OfflineAudioCompletionEvent.cpp',
+    'OscillatorNode.cpp',
+    'PannerNode.cpp',
+    'PeriodicWave.cpp',
+    'ScriptProcessorNode.cpp',
+    'StereoPannerNode.cpp',
+    'ThreeDPoint.cpp',
+    'WaveShaperNode.cpp',
+    'WebAudioUtils.cpp',
+]
+
+if CONFIG['CPU_ARCH'] == 'arm' and CONFIG['BUILD_ARM_NEON']:
+    SOURCES += ['AudioNodeEngineNEON.cpp']
+    SOURCES['AudioNodeEngineNEON.cpp'].flags += CONFIG['NEON_FLAGS']
+    LOCAL_INCLUDES += [
+        '/media/openmax_dl/dl/api/'
+    ]
+
+# Are we targeting x86 or x64?  If so, build SSE2 files.
+if CONFIG['INTEL_ARCHITECTURE']:
+    SOURCES += ['AudioNodeEngineSSE2.cpp']
+    DEFINES['USE_SSE2'] = True
+    SOURCES['AudioNodeEngineSSE2.cpp'].flags += CONFIG['SSE2_FLAGS']
+
+
+include('/ipc/chromium/chromium-config.mozbuild')
+
+FINAL_LIBRARY = 'xul'
+LOCAL_INCLUDES += [
+    '..'
+]
+
+if CONFIG['GNU_CXX']:
+    CXXFLAGS += ['-Wno-error=shadow']
diff --git a/dom/media/webaudio/test/audio-expected.wav b/dom/media/webaudio/test/audio-expected.wav
new file mode 100644
index 0000000000..1519270776
--- /dev/null
+++ b/dom/media/webaudio/test/audio-expected.wav
diff --git a/dom/media/webaudio/test/audio-mono-expected-2.wav b/dom/media/webaudio/test/audio-mono-expected-2.wav
new file mode 100644
index 0000000000..68c90dfa1e
--- /dev/null
+++ b/dom/media/webaudio/test/audio-mono-expected-2.wav
diff --git a/dom/media/webaudio/test/audio-mono-expected.wav b/dom/media/webaudio/test/audio-mono-expected.wav
new file mode 100644
index 0000000000..bf00e5cdf2
--- /dev/null
+++ b/dom/media/webaudio/test/audio-mono-expected.wav
diff --git a/dom/media/webaudio/test/audio-quad.wav b/dom/media/webaudio/test/audio-quad.wav
new file mode 100644
index 0000000000..093f0197ae
--- /dev/null
+++ b/dom/media/webaudio/test/audio-quad.wav
diff --git a/dom/media/webaudio/test/audio.ogv b/dom/media/webaudio/test/audio.ogv
new file mode 100644
index 0000000000..68dee3cf2b
--- /dev/null
+++ b/dom/media/webaudio/test/audio.ogv
diff --git a/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js b/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js
new file mode 100644
index 0000000000..2a5a4bff89
--- /dev/null
+++ b/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js
@@ -0,0 +1,3 @@
+onmessage = function (event) {
+  postMessage("Pong");
+};
diff --git a/dom/media/webaudio/test/audiovideo.mp4 b/dom/media/webaudio/test/audiovideo.mp4
new file mode 100644
index 0000000000..fe93122d29
--- /dev/null
+++ b/dom/media/webaudio/test/audiovideo.mp4
diff --git a/dom/media/webaudio/test/blink/README b/dom/media/webaudio/test/blink/README
new file mode 100644
index 0000000000..1d819221fd
--- /dev/null
+++ b/dom/media/webaudio/test/blink/README
@@ -0,0 +1,9 @@
+This directory contains tests originally borrowed from the Blink Web Audio test
+suite.
+
+The process of borrowing tests from Blink is as follows:
+
+* Import the pristine file from the Blink repo, noting the revision in the
+  commit message.
+* Modify the test files to turn the LayoutTest into a mochitest-plain and add
+* them to the test suite in a separate commit.
diff --git a/dom/media/webaudio/test/blink/audio-testing.js b/dom/media/webaudio/test/blink/audio-testing.js
new file mode 100644
index 0000000000..c66d32c7f2
--- /dev/null
+++ b/dom/media/webaudio/test/blink/audio-testing.js
@@ -0,0 +1,192 @@
+if (window.testRunner)
+    testRunner.overridePreference("WebKitWebAudioEnabled", "1");
+
+function writeString(s, a, offset) {
+    for (var i = 0; i < s.length; ++i) {
+        a[offset + i] = s.charCodeAt(i);
+    }
+}
+
+function writeInt16(n, a, offset) {
+    n = Math.floor(n);
+   
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+}
+
+function writeInt32(n, a, offset) {
+    n = Math.floor(n);
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+    var b3 = (n >> 16) & 255;
+    var b4 = (n >> 24) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+    a[offset + 2] = b3;
+    a[offset + 3] = b4;
+}
+
+function writeAudioBuffer(audioBuffer, a, offset) {
+    var n = audioBuffer.length;
+    var channels = audioBuffer.numberOfChannels;
+   
+    for (var i = 0; i < n; ++i) {
+        for (var k = 0; k < channels; ++k) {
+            var buffer = audioBuffer.getChannelData(k);
+            var sample = buffer[i] * 32768.0;
+
+            // Clip samples to the limitations of 16-bit.
+            // If we don't do this then we'll get nasty wrap-around distortion.
+            if (sample < -32768)
+                sample = -32768;
+            if (sample > 32767)
+                sample = 32767;
+
+            writeInt16(sample, a, offset);
+            offset += 2;
+        }
+    }
+}
+
+function createWaveFileData(audioBuffer) {
+    var frameLength = audioBuffer.length;
+    var numberOfChannels = audioBuffer.numberOfChannels;
+    var sampleRate = audioBuffer.sampleRate;
+    var bitsPerSample = 16;
+    var byteRate = sampleRate * numberOfChannels * bitsPerSample/8;
+    var blockAlign = numberOfChannels * bitsPerSample/8;
+    var wavDataByteLength = frameLength * numberOfChannels * 2; // 16-bit audio
+    var headerByteLength = 44;
+    var totalLength = headerByteLength + wavDataByteLength;
+
+    var waveFileData = new Uint8Array(totalLength);
+   
+    var subChunk1Size = 16; // for linear PCM
+    var subChunk2Size = wavDataByteLength;
+    var chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
+
+    writeString("RIFF", waveFileData, 0);
+    writeInt32(chunkSize, waveFileData, 4);
+    writeString("WAVE", waveFileData, 8);
+    writeString("fmt ", waveFileData, 12);
+   
+    writeInt32(subChunk1Size, waveFileData, 16);      // SubChunk1Size (4)
+    writeInt16(1, waveFileData, 20);                  // AudioFormat (2)
+    writeInt16(numberOfChannels, waveFileData, 22);   // NumChannels (2)
+    writeInt32(sampleRate, waveFileData, 24);         // SampleRate (4)
+    writeInt32(byteRate, waveFileData, 28);           // ByteRate (4)
+    writeInt16(blockAlign, waveFileData, 32);         // BlockAlign (2)
+    writeInt32(bitsPerSample, waveFileData, 34);      // BitsPerSample (4)
+                                                     
+    writeString("data", waveFileData, 36);            
+    writeInt32(subChunk2Size, waveFileData, 40);      // SubChunk2Size (4)
+   
+    // Write actual audio data starting at offset 44.
+    writeAudioBuffer(audioBuffer, waveFileData, 44);
+   
+    return waveFileData;
+}
+
+function createAudioData(audioBuffer) {
+    return createWaveFileData(audioBuffer);
+}
+
+function finishAudioTest(event) {
+    var audioData = createAudioData(event.renderedBuffer);
+    testRunner.setAudioData(audioData);
+    testRunner.notifyDone();
+}
+
+// Create an impulse in a buffer of length sampleFrameLength
+function createImpulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var k = 0; k < n; ++k) {
+        dataL[k] = 0;
+    }
+    dataL[0] = 1;
+
+    return audioBuffer;
+}
+
+// Create a buffer of the given length with a linear ramp having values 0 <= x < 1.
+function createLinearRampBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        dataL[i] = i / n;
+
+    return audioBuffer;
+}
+
+// Create a buffer of the given length having a constant value.
+function createConstantBuffer(context, sampleFrameLength, constantValue) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        dataL[i] = constantValue;
+
+    return audioBuffer;
+}
+
+// Create a stereo impulse in a buffer of length sampleFrameLength
+function createStereoImpulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(2, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+    var dataR = audioBuffer.getChannelData(1);
+
+    for (var k = 0; k < n; ++k) {
+        dataL[k] = 0;
+        dataR[k] = 0;
+    }
+    dataL[0] = 1;
+    dataR[0] = 1;
+
+    return audioBuffer;
+}
+
+// Convert time (in seconds) to sample frames.
+function timeToSampleFrame(time, sampleRate) {
+    return Math.floor(0.5 + time * sampleRate);
+}
+
+// Compute the number of sample frames consumed by start with
+// the specified |grainOffset|, |duration|, and |sampleRate|.
+function grainLengthInSampleFrames(grainOffset, duration, sampleRate) {
+    var startFrame = timeToSampleFrame(grainOffset, sampleRate);
+    var endFrame = timeToSampleFrame(grainOffset + duration, sampleRate);
+
+    return endFrame - startFrame;
+}
+
+// True if the number is not an infinity or NaN
+function isValidNumber(x) {
+    return !isNaN(x) && (x != Infinity) && (x != -Infinity);
+}
+
+function shouldThrowTypeError(func, text) {
+    var ok = false;
+    try {
+        func();
+    } catch (e) {
+        if (e instanceof TypeError) {
+            ok = true;
+        }
+    }
+    if (ok) {
+        testPassed(text + " threw TypeError.");
+    } else {
+        testFailed(text + " should throw TypeError.");
+    }
+}
diff --git a/dom/media/webaudio/test/blink/biquad-filters.js b/dom/media/webaudio/test/blink/biquad-filters.js
new file mode 100644
index 0000000000..06fff98b18
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-filters.js
@@ -0,0 +1,368 @@
+// Taken from WebKit/LayoutTests/webaudio/resources/biquad-filters.js
+
+// A biquad filter has a z-transform of
+// H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+//
+// The formulas for the various filters were taken from
+// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt.
+
+
+// Lowpass filter.
+function createLowpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+
+    if (freq == 1) {
+        // The formula below works, except for roundoff.  When freq = 1,
+        // the filter is just a wire, so hardwire the coefficients.
+        b0 = 1;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+        var cos_w0 = Math.cos(w0);
+
+        b0 = 0.5 * (1 - cos_w0);
+        b1 = 1 - cos_w0;
+        b2 = b0;
+        a0 = 1 + alpha;
+        a1 = -2.0 * cos_w0;
+        a2 = 1 - alpha;
+    }
+
+    return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function createHighpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a1;
+    var a2;
+
+    if (freq == 1) {
+        // The filter is 0
+        b0 = 0;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else if (freq == 0) {
+        // The filter is 1.  Computation of coefficients below is ok, but
+        // there's a pole at 1 and a zero at 1, so round-off could make
+        // the filter unstable.
+        b0 = 1;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+        var cos_w0 = Math.cos(w0);
+
+        b0 = 0.5 * (1 + cos_w0);
+        b1 = -1 - cos_w0;
+        b2 = b0;
+        a0 = 1 + alpha;
+        a1 = -2.0 * cos_w0;
+        a2 = 1 - alpha;
+    }
+
+    return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2) {
+    var scale = 1 / a0;
+
+    return {b0 : b0 * scale,
+            b1 : b1 * scale,
+            b2 : b2 * scale,
+            a1 : a1 * scale,
+            a2 : a2 * scale};
+}
+
+function createBandpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        var w0 = Math.PI * freq;
+        if (q > 0) {
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = alpha;
+            b1 = 0;
+            b2 = -alpha;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0, and frequency is not 0 or 1.  The above formula has a
+            // divide by zero problem.  The limit of the z-transform as q
+            // approaches 0 is 1, so set the filter that way.
+            coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // When freq = 0 or 1, the z-transform is identically 0,
+        // independent of q.
+        coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0}
+    }
+
+    return coef;
+}
+
+function createLowShelfFilter(freq, q, gain) {
+    // q not used
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var S = 1;
+    var A = Math.pow(10, gain / 40);
+
+    if (freq == 1) {
+        // The filter is just a constant gain
+        coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else if (freq == 0) {
+        // The filter is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 1 / 2 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        var k = Math.cos(w0);
+        var k2 = 2 * Math.sqrt(A) * alpha;
+        var Ap1 = A + 1;
+        var Am1 = A - 1;
+
+        b0 = A * (Ap1 - Am1 * k + k2);
+        b1 = 2 * A * (Am1 - Ap1 * k);
+        b2 = A * (Ap1 - Am1 * k - k2);
+        a0 = Ap1 + Am1 * k + k2;
+        a1 = -2 * (Am1 + Ap1 * k);
+        a2 = Ap1 + Am1 * k - k2;
+        coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+    }
+
+    return coef;
+}
+
+function createHighShelfFilter(freq, q, gain) {
+    // q not used
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var A = Math.pow(10, gain / 40);
+
+    if (freq == 1) {
+        // When freq = 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else if (freq > 0) {
+        var w0 = Math.PI * freq;
+        var S = 1;
+        var alpha = 0.5 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        var k = Math.cos(w0);
+        var k2 = 2 * Math.sqrt(A) * alpha;
+        var Ap1 = A + 1;
+        var Am1 = A - 1;
+
+        b0 = A * (Ap1 + Am1 * k + k2);
+        b1 = -2 * A * (Am1 + Ap1 * k);
+        b2 = A * (Ap1 + Am1 * k - k2);
+        a0 = Ap1 - Am1 * k + k2;
+        a1 = 2 * (Am1 - Ap1*k);
+        a2 = Ap1 - Am1 * k-k2;
+
+        coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+        // When freq = 0, the filter is just a gain
+        coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createPeakingFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var A = Math.pow(10, gain / 40);
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1 + alpha * A;
+            b1 = -2 * k;
+            b2 = 1 - alpha * A;
+            a0 = 1 + alpha / A;
+            a1 = -2 * k;
+            a2 = 1 - alpha / A;
+
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0, we have a divide by zero problem in the formulas
+            // above.  But if we look at the z-transform, we see that the
+            // limit as q approaches 0 is A^2.
+            coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // freq = 0 or 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createNotchFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1;
+            b1 = -2 * k;
+            b2 = 1;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When q = 0, we get a divide by zero above.  The limit of the
+            // z-transform as q approaches 0 is 0, so set the coefficients
+            // appropriately.
+            coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // When freq = 0 or 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createAllpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1 - alpha;
+            b1 = -2 * k;
+            b2 = 1 + alpha;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0
+            coef = {b0 : -1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function filterData(filterCoef, signal, len) {
+    var y = new Array(len);
+    var b0 = filterCoef.b0;
+    var b1 = filterCoef.b1;
+    var b2 = filterCoef.b2;
+    var a1 = filterCoef.a1;
+    var a2 = filterCoef.a2;
+
+    // Prime the pump. (Assumes the signal has length >= 2!)
+    y[0] = b0 * signal[0];
+    y[1] = b0 * signal[1] + b1 * signal[0] - a1 * y[0];
+
+    // Filter all of the signal that we have.
+    for (var k = 2; k < Math.min(signal.length, len); ++k) {
+        y[k] = b0 * signal[k] + b1 * signal[k-1] + b2 * signal[k-2] - a1 * y[k-1] - a2 * y[k-2];
+    }
+
+    // If we need to filter more, but don't have any signal left,
+    // assume the signal is zero.
+    for (var k = signal.length; k < len; ++k) {
+        y[k] = - a1 * y[k-1] - a2 * y[k-2];
+    }
+
+    return y;
+}
+
+// Map the filter type name to a function that computes the filter coefficents for the given filter
+// type.
+var filterCreatorFunction = {"lowpass": createLowpassFilter,
+                             "highpass": createHighpassFilter,
+                             "bandpass": createBandpassFilter,
+                             "lowshelf": createLowShelfFilter,
+                             "highshelf": createHighShelfFilter,
+                             "peaking": createPeakingFilter,
+                             "notch": createNotchFilter,
+                             "allpass": createAllpassFilter};
+
+var filterTypeName = {"lowpass": "Lowpass filter",
+                      "highpass": "Highpass filter",
+                      "bandpass": "Bandpass filter",
+                      "lowshelf": "Lowshelf filter",
+                      "highshelf": "Highshelf filter",
+                      "peaking": "Peaking filter",
+                      "notch": "Notch filter",
+                      "allpass": "Allpass filter"};
+
+function createFilter(filterType, freq, q, gain) {
+    return filterCreatorFunction[filterType](freq, q, gain);
+}
diff --git a/dom/media/webaudio/test/blink/biquad-testing.js b/dom/media/webaudio/test/blink/biquad-testing.js
new file mode 100644
index 0000000000..795adf6012
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-testing.js
@@ -0,0 +1,153 @@
+// Globals, to make testing and debugging easier.
+var context;
+var filter;
+var signal;
+var renderedBuffer;
+var renderedData;
+
+var sampleRate = 44100.0;
+var pulseLengthFrames = .1 * sampleRate;
+
+// Maximum allowed error for the test to succeed.  Experimentally determined.
+var maxAllowedError = 5.9e-8;
+
+// This must be large enough so that the filtered result is
+// essentially zero.  See comments for createTestAndRun.
+var timeStep = .1;
+
+// Maximum number of filters we can process (mostly for setting the
+// render length correctly.)
+var maxFilters = 5;
+
+// How long to render.  Must be long enough for all of the filters we
+// want to test.
+var renderLengthSeconds = timeStep * (maxFilters + 1) ;
+
+var renderLengthSamples = Math.round(renderLengthSeconds * sampleRate);
+
+// Number of filters that will be processed.
+var nFilters;
+
+function createImpulseBuffer(context, length) {
+    var impulse = context.createBuffer(1, length, context.sampleRate);
+    var data = impulse.getChannelData(0);
+    for (var k = 1; k < data.length; ++k) {
+        data[k] = 0;
+    }
+    data[0] = 1;
+
+    return impulse;
+}
+
+
+function createTestAndRun(context, filterType, filterParameters) {
+    // To test the filters, we apply a signal (an impulse) to each of
+    // the specified filters, with each signal starting at a different
+    // time.  The output of the filters is summed together at the
+    // output.  Thus for filter k, the signal input to the filter
+    // starts at time k * timeStep.  For this to work well, timeStep
+    // must be large enough for the output of each filter to have
+    // decayed to zero with timeStep seconds.  That way the filter
+    // outputs don't interfere with each other.
+
+    nFilters = Math.min(filterParameters.length, maxFilters);
+
+    signal = new Array(nFilters);
+    filter = new Array(nFilters);
+
+    impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+    // Create all of the signal sources and filters that we need.
+    for (var k = 0; k < nFilters; ++k) {
+        signal[k] = context.createBufferSource();
+        signal[k].buffer = impulse;
+
+        filter[k] = context.createBiquadFilter();
+        filter[k].type = filterType;
+        filter[k].frequency.value = context.sampleRate / 2 * filterParameters[k].cutoff;
+        filter[k].detune.value = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+        filter[k].Q.value = filterParameters[k].q;
+        filter[k].gain.value = filterParameters[k].gain;
+
+        signal[k].connect(filter[k]);
+        filter[k].connect(context.destination);
+
+        signal[k].start(timeStep * k);
+    }
+
+    context.oncomplete = checkFilterResponse(filterType, filterParameters);
+    context.startRendering();
+}
+
+function addSignal(dest, src, destOffset) {
+    // Add src to dest at the given dest offset.
+    for (var k = destOffset, j = 0; k < dest.length, j < src.length; ++k, ++j) {
+        dest[k] += src[j];
+    }
+}
+
+function generateReference(filterType, filterParameters) {
+    var result = new Array(renderLengthSamples);
+    var data = new Array(renderLengthSamples);
+    // Initialize the result array and data.
+    for (var k = 0; k < result.length; ++k) {
+        result[k] = 0;
+        data[k] = 0;
+    }
+    // Make data an impulse.
+    data[0] = 1;
+
+    for (var k = 0; k < nFilters; ++k) {
+        // Filter an impulse
+        var detune = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+        var frequency = filterParameters[k].cutoff * Math.pow(2, detune / 1200); // Apply detune, converting from Cents.
+
+        var filterCoef = createFilter(filterType,
+                                      frequency,
+                                      filterParameters[k].q,
+                                      filterParameters[k].gain);
+        var y = filterData(filterCoef, data, renderLengthSamples);
+
+        // Accumulate this filtered data into the final output at the desired offset.
+        addSignal(result, y, timeToSampleFrame(timeStep * k, sampleRate));
+    }
+
+    return result;
+}
+
+function checkFilterResponse(filterType, filterParameters) {
+    return function(event) {
+        renderedBuffer = event.renderedBuffer;
+        renderedData = renderedBuffer.getChannelData(0);
+
+        reference = generateReference(filterType, filterParameters);
+
+        var len = Math.min(renderedData.length, reference.length);
+
+        var success = true;
+
+        // Maximum error between rendered data and expected data
+        var maxError = 0;
+
+        // Sample offset where the maximum error occurred.
+        var maxPosition = 0;
+
+        // Number of infinities or NaNs that occurred in the rendered data.
+        var invalidNumberCount = 0;
+
+        ok(nFilters == filterParameters.length, "Test wanted " + filterParameters.length + " filters but only " + maxFilters + " allowed.");
+
+        compareChannels(renderedData, reference, len, 0, 0, true);
+
+        // Check for bad numbers in the rendered output too.
+        // There shouldn't be any.
+        for (var k = 0; k < len; ++k) {
+            if (!isValidNumber(renderedData[k])) {
+                ++invalidNumberCount;
+            }
+        }
+
+        ok(invalidNumberCount == 0, "Rendered output has " + invalidNumberCount + " infinities or NaNs.");
+        SimpleTest.finish();
+    }
+}
diff --git a/dom/media/webaudio/test/blink/convolution-testing.js b/dom/media/webaudio/test/blink/convolution-testing.js
new file mode 100644
index 0000000000..98ff0c7756
--- /dev/null
+++ b/dom/media/webaudio/test/blink/convolution-testing.js
@@ -0,0 +1,182 @@
+var sampleRate = 44100.0;
+
+var renderLengthSeconds = 8;
+var pulseLengthSeconds = 1;
+var pulseLengthFrames = pulseLengthSeconds * sampleRate;
+
+function createSquarePulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+    var n = audioBuffer.length;
+    var data = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        data[i] = 1;
+
+    return audioBuffer;
+}
+
+// The triangle buffer holds the expected result of the convolution.
+// It linearly ramps up from 0 to its maximum value (at the center)
+// then linearly ramps down to 0.  The center value corresponds to the
+// point where the two square pulses overlap the most.
+function createTrianglePulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+    var n = audioBuffer.length;
+    var halfLength = n / 2;
+    var data = audioBuffer.getChannelData(0);
+    
+    for (var i = 0; i < halfLength; ++i)
+        data[i] = i + 1;
+
+    for (var i = halfLength; i < n; ++i)
+        data[i] = n - i - 1;
+
+    return audioBuffer;
+}
+
+function log10(x) {
+  return Math.log(x)/Math.LN10;
+}
+
+function linearToDecibel(x) {
+  return 20*log10(x);
+}
+
+// Verify that the rendered result is very close to the reference
+// triangular pulse.
+function checkTriangularPulse(rendered, reference) {
+    var match = true;
+    var maxDelta = 0;
+    var valueAtMaxDelta = 0;
+    var maxDeltaIndex = 0;
+
+    for (var i = 0; i < reference.length; ++i) {
+        var diff = rendered[i] - reference[i];
+        var x = Math.abs(diff);
+        if (x > maxDelta) {
+            maxDelta = x;
+            valueAtMaxDelta = reference[i];
+            maxDeltaIndex = i;
+        }
+    }
+
+    // allowedDeviationFraction was determined experimentally.  It
+    // is the threshold of the relative error at the maximum
+    // difference between the true triangular pulse and the
+    // rendered pulse.
+    var allowedDeviationDecibels = -129.4;
+    var maxDeviationDecibels = linearToDecibel(maxDelta / valueAtMaxDelta);
+
+    if (maxDeviationDecibels <= allowedDeviationDecibels) {
+        testPassed("Triangular portion of convolution is correct.");
+    } else {
+        testFailed("Triangular portion of convolution is not correct.  Max deviation = " + maxDeviationDecibels + " dB at " + maxDeltaIndex);
+        match = false;
+    }
+
+    return match;
+}        
+
+// Verify that the rendered data is close to zero for the first part
+// of the tail.
+function checkTail1(data, reference, breakpoint) {
+    var isZero = true;
+    var tail1Max = 0;
+
+    for (var i = reference.length; i < reference.length + breakpoint; ++i) {
+        var mag = Math.abs(data[i]);
+        if (mag > tail1Max) {
+            tail1Max = mag;
+        }
+    }
+
+    // Let's find the peak of the reference (even though we know a
+    // priori what it is).
+    var refMax = 0;
+    for (var i = 0; i < reference.length; ++i) {
+      refMax = Math.max(refMax, Math.abs(reference[i]));
+    }
+
+    // This threshold is experimentally determined by examining the
+    // value of tail1MaxDecibels.
+    var threshold1 = -129.7;
+
+    var tail1MaxDecibels = linearToDecibel(tail1Max/refMax);
+    if (tail1MaxDecibels <= threshold1) {
+        testPassed("First part of tail of convolution is sufficiently small.");
+    } else {
+        testFailed("First part of tail of convolution is not sufficiently small: " + tail1MaxDecibels + " dB");
+        isZero = false;
+    }
+
+    return isZero;
+}
+
+// Verify that the second part of the tail of the convolution is
+// exactly zero.
+function checkTail2(data, reference, breakpoint) {
+    var isZero = true;
+    var tail2Max = 0;
+    // For the second part of the tail, the maximum value should be
+    // exactly zero.
+    var threshold2 = 0;
+    for (var i = reference.length + breakpoint; i < data.length; ++i) {
+        if (Math.abs(data[i]) > 0) {
+            isZero = false; 
+            break;
+        }
+    }
+
+    if (isZero) {
+        testPassed("Rendered signal after tail of convolution is silent.");
+    } else {
+        testFailed("Rendered signal after tail of convolution should be silent.");
+    }
+
+    return isZero;
+}
+
+function checkConvolvedResult(trianglePulse) {
+    return function(event) {
+        var renderedBuffer = event.renderedBuffer;
+
+        var referenceData = trianglePulse.getChannelData(0);
+        var renderedData = renderedBuffer.getChannelData(0);
+    
+        var success = true;
+    
+        // Verify the triangular pulse is actually triangular.
+
+        success = success && checkTriangularPulse(renderedData, referenceData);
+        
+        // Make sure that portion after convolved portion is totally
+        // silent.  But round-off prevents this from being completely
+        // true.  At the end of the triangle, it should be close to
+        // zero.  If we go farther out, it should be even closer and
+        // eventually zero.
+
+        // For the tail of the convolution (where the result would be
+        // theoretically zero), we partition the tail into two
+        // parts.  The first is the at the beginning of the tail,
+        // where we tolerate a small but non-zero value.  The second part is
+        // farther along the tail where the result should be zero.
+        
+        // breakpoint is the point dividing the first two tail parts
+        // we're looking at.  Experimentally determined.
+        var breakpoint = 12800;
+
+        success = success && checkTail1(renderedData, referenceData, breakpoint);
+        
+        success = success && checkTail2(renderedData, referenceData, breakpoint);
+        
+        if (success) {
+            testPassed("Test signal was correctly convolved.");
+        } else {
+            testFailed("Test signal was not correctly convolved.");
+        }
+
+        finishJSTest();
+    }
+}
diff --git a/dom/media/webaudio/test/blink/mochitest.ini b/dom/media/webaudio/test/blink/mochitest.ini
new file mode 100644
index 0000000000..28bceb3a47
--- /dev/null
+++ b/dom/media/webaudio/test/blink/mochitest.ini
@@ -0,0 +1,23 @@
+[DEFAULT]
+tags=msg
+tags = webaudio
+subsuite = media
+support-files =
+  biquad-filters.js
+  biquad-testing.js
+  ../webaudio.js
+
+[test_biquadFilterNodeAllPass.html]
+[test_biquadFilterNodeAutomation.html]
+skip-if = true # Known problems with Biquad automation, e.g. Bug 1155709
+[test_biquadFilterNodeBandPass.html]
+[test_biquadFilterNodeGetFrequencyResponse.html]
+[test_biquadFilterNodeHighPass.html]
+[test_biquadFilterNodeHighShelf.html]
+[test_biquadFilterNodeLowPass.html]
+[test_biquadFilterNodeLowShelf.html]
+[test_biquadFilterNodeNotch.html]
+[test_biquadFilterNodePeaking.html]
+[test_biquadFilterNodeTail.html]
+[test_iirFilterNode.html]
+[test_iirFilterNodeGetFrequencyResponse.html]
diff --git a/dom/media/webaudio/test/blink/panner-model-testing.js b/dom/media/webaudio/test/blink/panner-model-testing.js
new file mode 100644
index 0000000000..45460e2768
--- /dev/null
+++ b/dom/media/webaudio/test/blink/panner-model-testing.js
@@ -0,0 +1,210 @@
+var sampleRate = 48000.0;
+
+var numberOfChannels = 1;
+
+// Time step when each panner node starts.
+var timeStep = 0.001;
+
+// Length of the impulse signal.
+var pulseLengthFrames = Math.round(timeStep * sampleRate);
+
+// How many panner nodes to create for the test
+var nodesToCreate = 100;
+
+// Be sure we render long enough for all of our nodes.
+var renderLengthSeconds = timeStep * (nodesToCreate + 1);
+
+// These are global mostly for debugging.
+var context;
+var impulse;
+var bufferSource;
+var panner;
+var position;
+var time;
+      
+var renderedBuffer;
+var renderedLeft;
+var renderedRight;
+
+function createGraph(context, nodeCount) {
+    bufferSource = new Array(nodeCount);
+    panner = new Array(nodeCount);
+    position = new Array(nodeCount);
+    time = new Array(nodeCount);
+    // Angle between panner locations.  (nodeCount - 1 because we want
+    // to include both 0 and 180 deg.
+    var angleStep = Math.PI / (nodeCount - 1);
+
+    if (numberOfChannels == 2) {
+        impulse = createStereoImpulseBuffer(context, pulseLengthFrames);
+    }
+    else
+        impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+    for (var k = 0; k < nodeCount; ++k) {
+        bufferSource[k] = context.createBufferSource();
+        bufferSource[k].buffer = impulse;
+
+        panner[k] = context.createPanner();
+        panner[k].panningModel = "equalpower";
+        panner[k].distanceModel = "linear";
+
+        var angle = angleStep * k;
+        position[k] = {angle : angle, x : Math.cos(angle), z : Math.sin(angle)};
+        panner[k].positionX.value = position[k].x;
+        panner[k].positionZ.value = position[k].z;
+
+        bufferSource[k].connect(panner[k]);
+        panner[k].connect(context.destination);
+
+        // Start the source
+        time[k] = k * timeStep;
+        bufferSource[k].start(time[k]);
+    }
+}
+
+function createTestAndRun(context, nodeCount, numberOfSourceChannels) {
+    numberOfChannels = numberOfSourceChannels;
+
+    createGraph(context, nodeCount);
+
+    context.oncomplete = checkResult;
+    context.startRendering();
+}
+
+// Map our position angle to the azimuth angle (in degrees).
+//
+// An angle of 0 corresponds to an azimuth of 90 deg; pi, to -90 deg.
+function angleToAzimuth(angle) {
+    return 90 - angle * 180 / Math.PI;
+}
+
+// The gain caused by the EQUALPOWER panning model
+function equalPowerGain(angle) {
+    var azimuth = angleToAzimuth(angle);
+
+    if (numberOfChannels == 1) {
+        var panPosition = (azimuth + 90) / 180;
+
+        var gainL = Math.cos(0.5 * Math.PI * panPosition);
+        var gainR = Math.sin(0.5 * Math.PI * panPosition);
+
+        return { left : gainL, right : gainR };
+    } else {
+        if (azimuth <= 0) {
+            var panPosition = (azimuth + 90) / 90;
+    
+            var gainL = 1 + Math.cos(0.5 * Math.PI * panPosition);
+            var gainR = Math.sin(0.5 * Math.PI * panPosition);
+    
+            return { left : gainL, right : gainR };
+        } else {
+            var panPosition = azimuth / 90;
+    
+            var gainL = Math.cos(0.5 * Math.PI * panPosition);
+            var gainR = 1 + Math.sin(0.5 * Math.PI * panPosition);
+    
+            return { left : gainL, right : gainR };
+        }
+    }
+}
+
+function checkResult(event) {
+    renderedBuffer = event.renderedBuffer;
+    renderedLeft = renderedBuffer.getChannelData(0);
+    renderedRight = renderedBuffer.getChannelData(1);
+
+    // The max error we allow between the rendered impulse and the
+    // expected value.  This value is experimentally determined.  Set
+    // to 0 to make the test fail to see what the actual error is.
+    var maxAllowedError = 1.3e-6;
+  
+    var success = true;
+
+    // Number of impulses found in the rendered result.
+    var impulseCount = 0;
+
+    // Max (relative) error and the index of the maxima for the left
+    // and right channels.
+    var maxErrorL = 0;
+    var maxErrorIndexL = 0;
+    var maxErrorR = 0;
+    var maxErrorIndexR = 0;
+
+    // Number of impulses that don't match our expected locations.
+    var timeCount = 0;
+
+    // Locations of where the impulses aren't at the expected locations.
+    var timeErrors = new Array();
+
+    for (var k = 0; k < renderedLeft.length; ++k) {
+        // We assume that the left and right channels start at the same instant.
+        if (renderedLeft[k] != 0 || renderedRight[k] != 0) {
+            // The expected gain for the left and right channels.
+            var pannerGain = equalPowerGain(position[impulseCount].angle);
+            var expectedL = pannerGain.left;
+            var expectedR = pannerGain.right;
+
+            // Absolute error in the gain.
+            var errorL = Math.abs(renderedLeft[k] - expectedL);
+            var errorR = Math.abs(renderedRight[k] - expectedR);
+
+            if (Math.abs(errorL) > maxErrorL) {
+                maxErrorL = Math.abs(errorL);
+                maxErrorIndexL = impulseCount;
+            }
+            if (Math.abs(errorR) > maxErrorR) {
+                maxErrorR = Math.abs(errorR);
+                maxErrorIndexR = impulseCount;
+            }
+
+            // Keep track of the impulses that didn't show up where we
+            // expected them to be.
+            var expectedOffset = timeToSampleFrame(time[impulseCount], sampleRate);
+            if (k != expectedOffset) {
+                timeErrors[timeCount] = { actual : k, expected : expectedOffset};
+                ++timeCount;
+            }
+            ++impulseCount;
+        }
+    }
+
+    if (impulseCount == nodesToCreate) {
+        testPassed("Number of impulses matches the number of panner nodes.");
+    } else {
+        testFailed("Number of impulses is incorrect.  (Found " + impulseCount + " but expected " + nodesToCreate + ")");
+        success = false;
+    }
+
+    if (timeErrors.length > 0) {
+        success = false;
+        testFailed(timeErrors.length + " timing errors found in " + nodesToCreate + " panner nodes.");
+        for (var k = 0; k < timeErrors.length; ++k) {
+            testFailed("Impulse at sample " + timeErrors[k].actual + " but expected " + timeErrors[k].expected);
+        }
+    } else {
+        testPassed("All impulses at expected offsets.");
+    }
+
+    if (maxErrorL <= maxAllowedError) {
+        testPassed("Left channel gain values are correct.");
+    } else {
+        testFailed("Left channel gain values are incorrect.  Max error = " + maxErrorL + " at time " + time[maxErrorIndexL] + " (threshold = " + maxAllowedError + ")");
+        success = false;
+    }
+
+    if (maxErrorR <= maxAllowedError) {
+        testPassed("Right channel gain values are correct.");
+    } else {
+        testFailed("Right channel gain values are incorrect.  Max error = " + maxErrorR + " at time " + time[maxErrorIndexR] + " (threshold = " + maxAllowedError + ")");
+        success = false;
+    }
+
+    if (success) {
+        testPassed("EqualPower panner test passed");
+    } else {
+        testFailed("EqualPower panner test failed");
+    }
+
+    finishJSTest();
+}
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
new file mode 100644
index 0000000000..266521c523
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 1 },
+                            {cutoff : 1,    q : 10, gain : 1 },
+                            {cutoff : .5,   q :  0, gain : 1 },
+                            {cutoff : 0.25, q : 10, gain : 1 },
+                           ];
+    createTestAndRun(context, "allpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
new file mode 100644
index 0000000000..08ce71ccef
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
@@ -0,0 +1,351 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  // Don't need to run these tests at high sampling rate, so just use a low one to reduce memory
+  // usage and complexity.
+  var sampleRate = 16000;
+
+  // How long to render for each test.
+  var renderDuration = 1;
+
+  // The definition of the linear ramp automation function.
+  function linearRamp(t, v0, v1, t0, t1) {
+    return v0 + (v1 - v0) * (t - t0) / (t1 - t0);
+  }
+
+  // Generate the filter coefficients for the specified filter using the given parameters for
+  // the given duration.  |filterTypeFunction| is a function that returns the filter
+  // coefficients for one set of parameters.  |parameters| is a property bag that contains the
+  // start and end values (as an array) for each of the biquad attributes.  The properties are
+  // |freq|, |Q|, |gain|, and |detune|.  |duration| is the number of seconds for which the
+  // coefficients are generated.
+  //
+  // A property bag with properties |b0|, |b1|, |b2|, |a1|, |a2|.  Each propery is an array
+  // consisting of the coefficients for the time-varying biquad filter.
+  function generateFilterCoefficients(filterTypeFunction, parameters, duration) {
+     var endFrame = Math.ceil(duration * sampleRate);
+     var nCoef = endFrame;
+     var b0 = new Float64Array(nCoef);
+     var b1 = new Float64Array(nCoef);
+     var b2 = new Float64Array(nCoef);
+     var a1 = new Float64Array(nCoef);
+     var a2 = new Float64Array(nCoef);
+
+     var k = 0;
+     // If the property is not given, use the defaults.
+     var freqs = parameters.freq || [350, 350];
+     var qs = parameters.Q || [1, 1];
+     var gains = parameters.gain || [0, 0];
+     var detunes = parameters.detune || [0, 0];
+
+     for (var frame = 0; frame < endFrame; ++frame) {
+        // Apply linear ramp at frame |frame|.
+        var f = linearRamp(frame / sampleRate, freqs[0], freqs[1], 0, duration);
+        var q = linearRamp(frame / sampleRate, qs[0], qs[1], 0, duration);
+        var g = linearRamp(frame / sampleRate, gains[0], gains[1], 0, duration);
+        var d = linearRamp(frame / sampleRate, detunes[0], detunes[1], 0, duration);
+
+        // Compute actual frequency parameter
+        f = f * Math.pow(2, d / 1200);
+
+        // Compute filter coefficients
+        var coef = filterTypeFunction(f / (sampleRate / 2), q, g);
+        b0[k] = coef.b0;
+        b1[k] = coef.b1;
+        b2[k] = coef.b2;
+        a1[k] = coef.a1;
+        a2[k] = coef.a2;
+        ++k;
+     }
+
+     return {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2};
+  }
+
+  // Apply the given time-varying biquad filter to the given signal, |signal|.  |coef| should be
+  // the time-varying coefficients of the filter, as returned by |generateFilterCoefficients|.
+  function timeVaryingFilter(signal, coef) {
+    var length = signal.length;
+    // Use double precision for the internal computations.
+    var y = new Float64Array(length);
+
+    // Prime the pump. (Assumes the signal has length >= 2!)
+    y[0] = coef.b0[0] * signal[0];
+    y[1] = coef.b0[1] * signal[1] + coef.b1[1] * signal[0] - coef.a1[1] * y[0];
+
+    for (var n = 2; n < length; ++n) {
+      y[n] = coef.b0[n] * signal[n] + coef.b1[n] * signal[n-1] + coef.b2[n] * signal[n-2];
+      y[n] -= coef.a1[n] * y[n-1] + coef.a2[n] * y[n-2];
+    }
+
+    // But convert the result to single precision for comparison.
+    return y.map(Math.fround);
+  }
+
+  // Configure the audio graph using |context|.  Returns the biquad filter node and the
+  // AudioBuffer used for the source.
+  function configureGraph(context, toneFrequency) {
+    // The source is just a simple sine wave.
+    var src = context.createBufferSource();
+    var b = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+    var data = b.getChannelData(0);
+    var omega = 2 * Math.PI * toneFrequency / sampleRate;
+    for (var k = 0; k < data.length; ++k) {
+      data[k] = Math.sin(omega * k);
+    }
+    src.buffer = b;
+    var f = context.createBiquadFilter();
+    src.connect(f);
+    f.connect(context.destination);
+
+    src.start();
+
+    return {filter: f, source: b};
+  }
+
+  function createFilterVerifier(filterCreator, threshold, parameters, input, message) {
+    return function (resultBuffer) {
+      var actual = resultBuffer.getChannelData(0);
+      var coefs = generateFilterCoefficients(filterCreator, parameters, renderDuration);
+
+      reference = timeVaryingFilter(input, coefs);
+
+      compareChannels(actual, reference);
+    };
+  }
+
+  var testPromises = [];
+
+  // Automate just the frequency parameter.  A bandpass filter is used where the center
+  // frequency is swept across the source (which is a simple tone).
+  testPromises.push(function () {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Center frequency of bandpass filter and also the frequency of the test tone.
+    var centerFreq = 10*440;
+
+    // Sweep the frequency +/- 9*440 Hz from the center.  This should cause the output to low at
+    // the beginning and end of the test where the done is outside the pass band of the filter,
+    // but high in the center where the tone is near the center of the pass band.
+    var parameters = {
+      freq: [centerFreq - 9*440, centerFreq + 9*440]
+    }
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.setValueAtTime(parameters.freq[0], 0);
+    f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+
+    return context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 5e-5, parameters, b.getChannelData(0),
+        "Output of bandpass filter with frequency automation"));
+  }());
+
+  // Automate just the Q parameter.  A bandpass filter is used where the Q of the filter is
+  // swept.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // The frequency of the test tone.
+    var centerFreq = 440;
+
+    // Sweep the Q paramter between 1 and 200.  This will cause the output of the filter to pass
+    // most of the tone at the beginning to passing less of the tone at the end.  This is
+    // because we set center frequency of the bandpass filter to be slightly off from the actual
+    // tone.
+    var parameters = {
+      Q: [1, 200],
+      // Center frequency of the bandpass filter is just 25 Hz above the tone frequency.
+      freq: [centerFreq + 25, centerFreq + 25]
+    };
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.value = parameters.freq[0];
+    f.Q.setValueAtTime(parameters.Q[0], 0);
+    f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+
+    return context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 1.4e-6, parameters, b.getChannelData(0),
+        "Output of bandpass filter with Q automation"));
+  }());
+
+  // Automate just the gain of the lowshelf filter.  A test tone will be in the lowshelf part of
+  // the filter.  The output will vary as the gain of the lowshelf is changed.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Frequency of the test tone.
+    var centerFreq = 440;
+
+    // Set the cutoff frequency of the lowshelf to be significantly higher than the test tone.
+    // Sweep the gain from 20 dB to -20 dB.  (We go from 20 to -20 to easily verify that the
+    // filter didn't go unstable.)
+    var parameters = {
+      freq: [3500, 3500],
+      gain: [20, -20]
+    }
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "lowshelf";
+    f.frequency.value = parameters.freq[0];
+    f.gain.setValueAtTime(parameters.gain[0], 0);
+    f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createLowShelfFilter, 8e-6, parameters, b.getChannelData(0),
+        "Output of lowshelf filter with gain automation"));
+  }());
+
+  // Automate just the detune parameter.  Basically the same test as for the frequncy parameter
+  // but we just use the detune parameter to modulate the frequency parameter.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+    var centerFreq = 10*440;
+    var parameters = {
+      freq: [centerFreq, centerFreq],
+      detune: [-10*1200, 10*1200]
+    };
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.value = parameters.freq[0];
+    f.detune.setValueAtTime(parameters.detune[0], 0);
+    f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 5e-6, parameters, b.getChannelData(0),
+        "Output of bandpass filter with detune automation"));
+  }());
+
+  // Automate all of the filter parameters at once.  This is a basic check that everything is
+  // working.  A peaking filter is used because it uses all of the parameters.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+    var graph = configureGraph(context, 10*440);
+    var f = graph.filter;
+    var b = graph.source;
+
+    // Sweep all of the filter parameters.  These are pretty much arbitrary.
+    var parameters = {
+      freq: [10000, 100],
+      Q: [f.Q.value, .0001],
+      gain: [f.gain.value, 20],
+      detune: [2400, -2400]
+    };
+
+    f.type = "peaking";
+    // Set starting points for all parameters of the filter.  Start at 10 kHz for the center
+    // frequency, and the defaults for Q and gain.
+    f.frequency.setValueAtTime(parameters.freq[0], 0);
+    f.Q.setValueAtTime(parameters.Q[0], 0);
+    f.gain.setValueAtTime(parameters.gain[0], 0);
+    f.detune.setValueAtTime(parameters.detune[0], 0);
+
+    // Linear ramp each parameter
+    f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+    f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+    f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+    f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createPeakingFilter, 3.3e-4, parameters, b.getChannelData(0),
+        "Output of peaking filter with automation of all parameters"));
+  }());
+
+  // Test that modulation of the frequency parameter of the filter works.  A sinusoid of 440 Hz
+  // is the test signal that is applied to a bandpass biquad filter.  The frequency parameter of
+  // the filter is modulated by a sinusoid at 103 Hz, and the frequency modulation varies from
+  // 116 to 412 Hz.  (This test was taken from the description in
+  // https://github.com/WebAudio/web-audio-api/issues/509#issuecomment-94731355)
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Create a graph with the sinusoidal source at 440 Hz as the input to a biquad filter.
+    var graph = configureGraph(context, 440);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.Q.value = 5;
+    f.frequency.value = 264;
+
+    // Create the modulation source, a sinusoid with frequency 103 Hz and amplitude 148.  (The
+    // amplitude of 148 is added to the filter's frequency value of 264 to produce a sinusoidal
+    // modulation of the frequency parameter from 116 to 412 Hz.)
+    var mod = context.createBufferSource();
+    var mbuffer = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+    var d = mbuffer.getChannelData(0);
+    var omega = 2 * Math.PI * 103 / sampleRate;
+    for (var k = 0; k < d.length; ++k) {
+      d[k] = 148 * Math.sin(omega * k);
+    }
+    mod.buffer = mbuffer;
+
+    mod.connect(f.frequency);
+
+    mod.start();
+    return context.startRendering()
+      .then(function (resultBuffer) {
+         var actual = resultBuffer.getChannelData(0);
+         // Compute the filter coefficients using the mod sine wave
+
+         var endFrame = Math.ceil(renderDuration * sampleRate);
+         var nCoef = endFrame;
+         var b0 = new Float64Array(nCoef);
+         var b1 = new Float64Array(nCoef);
+         var b2 = new Float64Array(nCoef);
+         var a1 = new Float64Array(nCoef);
+         var a2 = new Float64Array(nCoef);
+
+         // Generate the filter coefficients when the frequency varies from 116 to 248 Hz using
+         // the 103 Hz sinusoid.
+         for (var k = 0; k < nCoef; ++k) {
+           var freq = f.frequency.value + d[k];
+           var c = createBandpassFilter(freq / (sampleRate / 2), f.Q.value, f.gain.value);
+           b0[k] = c.b0;
+           b1[k] = c.b1;
+           b2[k] = c.b2;
+           a1[k] = c.a1;
+           a2[k] = c.a2;
+         }
+         reference = timeVaryingFilter(b.getChannelData(0),
+           {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2});
+
+         compareChannels(actual, reference);
+       });
+  }());
+
+  // Wait for all tests
+  Promise.all(testPromises).then(function () {
+    SimpleTest.finish();
+  }, function () {
+    SimpleTest.finish();
+  });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
new file mode 100644
index 0000000000..a3a1484f6f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Band Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 0, gain : 1 },
+                            {cutoff : 1,    q : 0, gain : 1 },
+                            {cutoff : 0.5,  q : 0, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "bandpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
new file mode 100644
index 0000000000..1576db1e8c
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,261 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+// Test the frequency response of a biquad filter.  We compute the frequency response for a simple
+// peaking biquad filter and compare it with the expected frequency response.  The actual filter
+// used doesn't matter since we're testing getFrequencyResponse and not the actual filter output.
+// The filters are extensively tested in other biquad tests.
+
+var context;
+
+// The biquad filter node.
+var filter;
+
+// The magnitude response of the biquad filter.
+var magResponse;
+
+// The phase response of the biquad filter.
+var phaseResponse;
+
+// Number of frequency samples to take.
+var numberOfFrequencies = 1000;
+
+// The filter parameters.
+var filterCutoff = 1000; // Hz.
+var filterQ = 1;
+var filterGain = 5; // Decibels.
+
+// The maximum allowed error in the magnitude response.
+var maxAllowedMagError = 5.7e-7;
+
+// The maximum allowed error in the phase response.
+var maxAllowedPhaseError = 4.7e-8;
+
+// The magnitudes and phases of the reference frequency response.
+var magResponse;
+var phaseResponse;
+
+// The magnitudes and phases of the reference frequency response.
+var expectedMagnitudes;
+var expectedPhases;
+
+// Convert frequency in Hz to a normalized frequency between 0 to 1 with 1 corresponding to the
+// Nyquist frequency.
+function normalizedFrequency(freqHz, sampleRate)
+{
+    var nyquist = sampleRate / 2;
+    return freqHz / nyquist;
+}
+
+// Get the filter response at a (normalized) frequency |f| for the filter with coefficients |coef|.
+function getResponseAt(coef, f)
+{
+    var b0 = coef.b0;
+    var b1 = coef.b1;
+    var b2 = coef.b2;
+    var a1 = coef.a1;
+    var a2 = coef.a2;
+
+    // H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+    //
+    // Compute H(exp(i * pi * f)).  No native complex numbers in javascript, so break H(exp(i * pi * // f))
+    // in to the real and imaginary parts of the numerator and denominator.  Let omega = pi * f.
+    // Then the numerator is
+    //
+    // b0 + b1 * cos(omega) + b2 * cos(2 * omega) - i * (b1 * sin(omega) + b2 * sin(2 * omega))
+    //
+    // and the denominator is
+    //
+    // 1 + a1 * cos(omega) + a2 * cos(2 * omega) - i * (a1 * sin(omega) + a2 * sin(2 * omega))
+    //
+    // Compute the magnitude and phase from the real and imaginary parts.
+
+    var omega = Math.PI * f;
+    var numeratorReal = b0 + b1 * Math.cos(omega) + b2 * Math.cos(2 * omega);
+    var numeratorImag = -(b1 * Math.sin(omega) + b2 * Math.sin(2 * omega));
+    var denominatorReal = 1 + a1 * Math.cos(omega) + a2 * Math.cos(2 * omega);
+    var denominatorImag = -(a1 * Math.sin(omega) + a2 * Math.sin(2 * omega));
+
+    var magnitude = Math.sqrt((numeratorReal * numeratorReal + numeratorImag * numeratorImag)
+                              / (denominatorReal * denominatorReal + denominatorImag * denominatorImag));
+    var phase = Math.atan2(numeratorImag, numeratorReal) - Math.atan2(denominatorImag, denominatorReal);
+
+    if (phase >= Math.PI) {
+        phase -= 2 * Math.PI;
+    } else if (phase <= -Math.PI) {
+        phase += 2 * Math.PI;
+    }
+
+    return {magnitude : magnitude, phase : phase};
+}
+
+// Compute the reference frequency response for the biquad filter |filter| at the frequency samples
+// given by |frequencies|.
+function frequencyResponseReference(filter, frequencies)
+{
+    var sampleRate = filter.context.sampleRate;
+    var normalizedFreq = normalizedFrequency(filter.frequency.value, sampleRate);
+    var filterCoefficients = createFilter(filter.type, normalizedFreq, filter.Q.value, filter.gain.value);
+
+    var magnitudes = [];
+    var phases = [];
+
+    for (var k = 0; k < frequencies.length; ++k) {
+        var response = getResponseAt(filterCoefficients, normalizedFrequency(frequencies[k], sampleRate));
+        magnitudes.push(response.magnitude);
+        phases.push(response.phase);
+    }
+
+    return {magnitudes : magnitudes, phases : phases};
+}
+
+// Compute a set of linearly spaced frequencies.
+function createFrequencies(nFrequencies, sampleRate)
+{
+    var frequencies = new Float32Array(nFrequencies);
+    var nyquist = sampleRate / 2;
+    var freqDelta = nyquist / nFrequencies;
+
+    for (var k = 0; k < nFrequencies; ++k) {
+        frequencies[k] = k * freqDelta;
+    }
+
+    return frequencies;
+}
+
+function linearToDecibels(x)
+{
+    if (x) {
+        return 20 * Math.log(x) / Math.LN10;
+    } else {
+        return -1000;
+    }
+}
+
+// Look through the array and find any NaN or infinity. Returns the index of the first occurence or
+// -1 if none.
+function findBadNumber(signal)
+{
+    for (var k = 0; k < signal.length; ++k) {
+        if (!isValidNumber(signal[k])) {
+           return k;
+        }
+    }
+    return -1;
+}
+
+// Compute absolute value of the difference between phase angles, taking into account the wrapping
+// of phases.
+function absolutePhaseDifference(x, y)
+{
+    var diff = Math.abs(x - y);
+
+    if (diff > Math.PI) {
+        diff = 2 * Math.PI - diff;
+    }
+    return diff;
+}
+
+// Compare the frequency response with our expected response.
+function compareResponses(filter, frequencies, magResponse, phaseResponse)
+{
+    var expectedResponse = frequencyResponseReference(filter, frequencies);
+
+    expectedMagnitudes = expectedResponse.magnitudes;
+    expectedPhases = expectedResponse.phases;
+
+    var n = magResponse.length;
+    var success = true;
+    var badResponse = false;
+
+    var maxMagError = -1;
+    var maxMagErrorIndex = -1;
+
+    var k;
+    var hasBadNumber;
+
+    hasBadNumber = findBadNumber(magResponse);
+    ok (hasBadNumber < 0, "Magnitude response has NaN or infinity at " + hasBadNumber);
+
+    hasBadNumber = findBadNumber(phaseResponse);
+    ok (hasBadNumber < 0, "Phase response has NaN or infinity at " + hasBadNumber);
+
+    // These aren't testing the implementation itself.  Instead, these are sanity checks on the
+    // reference.  Failure here does not imply an error in the implementation.
+    hasBadNumber = findBadNumber(expectedMagnitudes);
+    ok (hasBadNumber < 0, "Expected magnitude response has NaN or infinity at " + hasBadNumber);
+
+    hasBadNumber = findBadNumber(expectedPhases);
+    ok (hasBadNumber < 0, "Expected phase response has NaN or infinity at " + hasBadNumber);
+
+    for (k = 0; k < n; ++k) {
+        var error = Math.abs(linearToDecibels(magResponse[k]) - linearToDecibels(expectedMagnitudes[k]));
+        if (error > maxMagError) {
+            maxMagError = error;
+            maxMagErrorIndex = k;
+        }
+    }
+
+    var message = "Magnitude error (" + maxMagError + " dB)";
+    message += " exceeded threshold at " + frequencies[maxMagErrorIndex];
+    message += " Hz.  Actual: " + linearToDecibels(magResponse[maxMagErrorIndex]);
+    message += " dB, expected: " + linearToDecibels(expectedMagnitudes[maxMagErrorIndex]) + " dB.";
+    ok(maxMagError < maxAllowedMagError, message);
+
+    var maxPhaseError = -1;
+    var maxPhaseErrorIndex = -1;
+
+    for (k = 0; k < n; ++k) {
+        var error = absolutePhaseDifference(phaseResponse[k], expectedPhases[k]);
+        if (error > maxPhaseError) {
+            maxPhaseError = error;
+            maxPhaseErrorIndex = k;
+        }
+    }
+
+    message = "Phase error (radians) (" + maxPhaseError;
+    message += ") exceeded threshold at " + frequencies[maxPhaseErrorIndex];
+    message += " Hz.  Actual: " + phaseResponse[maxPhaseErrorIndex];
+    message += " expected: " + expectedPhases[maxPhaseErrorIndex];
+
+    ok(maxPhaseError < maxAllowedPhaseError, message);
+}
+
+context = new AudioContext();
+
+filter = context.createBiquadFilter();
+
+// Arbitrarily test a peaking filter, but any kind of filter can be tested.
+filter.type = "peaking";
+filter.frequency.value = filterCutoff;
+filter.Q.value = filterQ;
+filter.gain.value = filterGain;
+
+var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+magResponse = new Float32Array(numberOfFrequencies);
+phaseResponse = new Float32Array(numberOfFrequencies);
+
+filter.getFrequencyResponse(frequencies, magResponse, phaseResponse);
+compareResponses(filter, frequencies, magResponse, phaseResponse);
+
+SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
new file mode 100644
index 0000000000..cb9aa274c7
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode High Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 1, gain : 1 },
+                            {cutoff : 1,    q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "highpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
new file mode 100644
index 0000000000..3581459b0f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode High Shelf Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "highshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
new file mode 100644
index 0000000000..b0c12558f6
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 1, gain : 1 },
+                            {cutoff : 1,    q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1, detune : 100 },
+                            {cutoff : 0.01, q : 1, gain : 1, detune : -200 },
+                           ];
+    createTestAndRun(context, "lowpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
new file mode 100644
index 0000000000..3c83bfaa39
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Shelf Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ 
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "lowshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
new file mode 100644
index 0000000000..551410c669
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Notch Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 1 },
+                            {cutoff : 1,    q : 10, gain : 1 },
+                            {cutoff : .5,   q :  0, gain : 1 },
+                            {cutoff : 0.25, q : 10, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "notch", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
new file mode 100644
index 0000000000..33fcc225a7
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : .5,   q :  0, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "peaking", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
new file mode 100644
index 0000000000..fd02e734fa
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  // A high sample rate shows the issue more clearly.
+  var sampleRate = 192000;
+  // Some short duration because we don't need to run the test for very long.
+  var testDurationSec = 0.5;
+  var testDurationFrames = testDurationSec * sampleRate;
+
+  // Amplitude experimentally determined to give a biquad output close to 1. (No attempt was
+  // made to produce exactly 1; it's not needed.)
+  var sourceAmplitude = 100;
+
+  // The output of the biquad filter should not change by more than this much between output
+  // samples.  Threshold was determined experimentally.
+  var glitchThreshold = 0.01292;
+
+  // Test that a Biquad filter doesn't have it's output terminated because the input has gone
+  // away.  Generally, when a source node is finished, it disconnects itself from any downstream
+  // nodes.  This is the correct behavior.  Nodes that have no inputs (disconnected) are
+  // generally assumed to output zeroes.  This is also desired behavior.  However, biquad
+  // filters have memory so they should not suddenly output zeroes when the input is
+  // disconnected.  This test checks to see if the output doesn't suddenly change to zero.
+  var context = new OfflineAudioContext(1, testDurationFrames, sampleRate);
+
+  // Create an impulse source.
+  var buffer = context.createBuffer(1, 1, context.sampleRate);
+  buffer.getChannelData(0)[0] = sourceAmplitude;
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+
+  // Create the biquad filter. It doesn't really matter what kind, so the default filter type
+  // and parameters is fine.  Connect the source to it.
+  var biquad = context.createBiquadFilter();
+  source.connect(biquad);
+  biquad.connect(context.destination);
+
+  source.start();
+
+  context.startRendering().then(function(result) {
+    // There should be no large discontinuities in the output
+    var buffer = result.getChannelData(0);
+    var maxGlitchIndex = 0;
+    var maxGlitchValue = 0.0;
+    for (var i = 1; i < buffer.length; i++) {
+      var diff = Math.abs(buffer[i-1] - buffer[i]);
+      if (diff >= glitchThreshold) {
+        if (diff > maxGlitchValue) {
+          maxGlitchIndex = i;
+          maxGlitchValue = diff;
+        }
+      }
+    }
+    ok(maxGlitchIndex == 0, 'glitches detected in biquad output: maximum glitch at ' + maxGlitchIndex + ' with diff of ' + maxGlitchValue);
+    SimpleTest.finish();
+  })
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNode.html b/dom/media/webaudio/test/blink/test_iirFilterNode.html
new file mode 100644
index 0000000000..47f936761f
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNode.html
@@ -0,0 +1,467 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode GetFrequencyResponse</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <script type="text/javascript" src="biquad-filters.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var sampleRate = 48000;
+  var testDurationSec = 1;
+  var testFrames = testDurationSec * sampleRate;
+
+  var testPromises = []
+  testPromises.push(function () {
+    // Test that the feedback coefficients are normalized.  Do this be creating two
+    // IIRFilterNodes.  One has normalized coefficients, and one doesn't.  Compute the
+    // difference and make sure they're the same.
+    var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+    // Use a simple impulse as the source.
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Gain node for computing the difference between the filters.
+    var gain = context.createGain();
+    gain.gain.value = -1;
+
+    // The IIR filters.  Use a common feedforward array.
+    var ff = [1];
+
+    var fb1 = [1, .9];
+
+    var fb2 = new Float64Array(2);
+    // Scale the feedback coefficients by an arbitrary factor.
+    var coefScaleFactor = 2;
+    for (var k = 0; k < fb2.length; ++k) {
+      fb2[k] = coefScaleFactor * fb1[k];
+    }
+
+    var iir1 = context.createIIRFilter(ff, fb1);
+    var iir2 = context.createIIRFilter(ff, fb2);
+
+    // Create the graph.  The output of iir1 (normalized coefficients) is channel 0, and the
+    // output of iir2 (unnormalized coefficients), with appropriate scaling, is channel 1.
+    var merger = context.createChannelMerger(2);
+    source.connect(iir1);
+    source.connect(iir2);
+    iir1.connect(merger, 0, 0);
+    iir2.connect(gain);
+
+    // The gain for the gain node should be set to compensate for the scaling of the
+    // coefficients.  Since iir2 has scaled the coefficients by coefScaleFactor, the output is
+    // reduced by the same factor, so adjust the gain to scale the output of iir2 back up.
+    gain.gain.value = coefScaleFactor;
+    gain.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    source.start();
+
+    // Rock and roll!
+
+    return context.startRendering().then(function (result) {
+      // Find the max amplitude of the result, which should be near zero.
+      var iir1Data = result.getChannelData(0);
+      var iir2Data = result.getChannelData(1);
+
+      // Threshold isn't exactly zero because the arithmetic is done differently between the
+      // IIRFilterNode and the BiquadFilterNode.
+      compareChannels(iir1Data, iir2Data);
+    });
+  }());
+
+  testPromises.push(function () {
+    // Create a simple 1-zero filter and compare with the expected output.
+    var context = new OfflineAudioContext(1, testFrames, sampleRate);
+
+    // Use a simple impulse as the source
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // The filter is y(n) = 0.5*(x(n) + x(n-1)), a simple 2-point moving average.  This is
+    // rather arbitrary; keep it simple.
+
+    var iir = context.createIIRFilter([0.5, 0.5], [1]);
+
+    // Create the graph
+    source.connect(iir);
+    iir.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      var actual = result.getChannelData(0);
+      var expected = new Float64Array(testFrames);
+      // The filter is a simple 2-point moving average of an impulse, so the first two values
+      // are non-zero and the rest are zero.
+      expected[0] = 0.5;
+      expected[1] = 0.5;
+      compareChannels(actual, expected);
+    });
+  }());
+
+  testPromises.push(function () {
+    // Create a simple 1-pole filter and compare with the expected output.
+
+    // The filter is y(n) + c*y(n-1)= x(n).  The analytical response is (-c)^n, so choose a
+    // suitable number of frames to run the test for where the output isn't flushed to zero.
+    var c = 0.9;
+    var eps = 1e-20;
+    var duration = Math.floor(Math.log(eps) / Math.log(Math.abs(c)));
+    var context = new OfflineAudioContext(1, duration, sampleRate);
+
+    // Use a simple impulse as the source
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    var iir = context.createIIRFilter([1], [1, c]);
+
+    // Create the graph
+    source.connect(iir);
+    iir.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      var actual = result.getChannelData(0);
+      var expected = new Float64Array(actual.length);
+
+      // The filter is a simple 1-pole filter: y(n) = -c*y(n-k)+x(n), with an impulse as the
+      // input.
+      expected[0] = 1;
+      for (k = 1; k < testFrames; ++k) {
+        expected[k] = -c * expected[k-1];
+      }
+
+      compareChannels(actual, expected);
+    });
+  }());
+
+  // This function creates an IIRFilterNode equivalent to the specified
+  // BiquadFilterNode and compares the outputs.  The
+  // outputs from the two filters should be virtually identical.
+  function testWithBiquadFilter(filterType) {
+    var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+    // Use a constant (step function) as the source
+    var buffer = context.createBuffer(1, testFrames, context.sampleRate);
+    for (var i = 0; i < testFrames; ++i) {
+      buffer.getChannelData(0)[i] = 1;
+    }
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Create the biquad.  Choose some rather arbitrary values for Q and gain for the biquad
+    // so that the shelf filters aren't identical.
+    var biquad = context.createBiquadFilter();
+    biquad.type = filterType;
+    biquad.Q.value = 10;
+    biquad.gain.value = 10;
+
+    // Create the equivalent IIR Filter node by computing the coefficients of the given biquad
+    // filter type.
+    var nyquist = sampleRate / 2;
+    var coef = createFilter(filterType,
+                            biquad.frequency.value / nyquist,
+                            biquad.Q.value,
+                            biquad.gain.value);
+
+    var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+    var merger = context.createChannelMerger(2);
+    // Create the graph
+    source.connect(biquad);
+    source.connect(iir);
+
+    biquad.connect(merger, 0, 0);
+    iir.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      // Find the max amplitude of the result, which should be near zero.
+      var expected = result.getChannelData(0);
+      var actual = result.getChannelData(1);
+      compareChannels(actual, expected);
+    });
+  }
+
+  biquadFilterTypes = ["lowpass", "highpass", "bandpass", "notch",
+                       "allpass", "lowshelf", "highshelf", "peaking"];
+
+  // Create a set of tasks based on biquadTestConfigs.
+  for (var i = 0; i < biquadFilterTypes.length; ++i) {
+    testPromises.push(testWithBiquadFilter(biquadFilterTypes[i]));
+  }
+
+  testPromises.push(function () {
+    // Multi-channel test.  Create a biquad filter and the equivalent IIR filter.  Filter the
+    // same multichannel signal and compare the results.
+    var nChannels = 3;
+    var context = new OfflineAudioContext(nChannels, testFrames, sampleRate);
+
+    // Create a set of oscillators as the multi-channel source.
+    var source = [];
+
+    for (k = 0; k < nChannels; ++k) {
+      source[k] = context.createOscillator();
+      source[k].type = "sawtooth";
+      // The frequency of the oscillator is pretty arbitrary, but each oscillator should have a
+      // different frequency.
+      source[k].frequency.value = 100 + k * 100;
+    }
+
+    var merger = context.createChannelMerger(3);
+
+    var biquad = context.createBiquadFilter();
+
+    // Create the equivalent IIR Filter node.
+    var nyquist = sampleRate / 2;
+    var coef = createFilter(biquad.type,
+      biquad.frequency.value / nyquist,
+      biquad.Q.value,
+      biquad.gain.value);
+    var fb = [1, coef.a1, coef.a2];
+    var ff = [coef.b0, coef.b1, coef.b2];
+
+    var iir = context.createIIRFilter(ff, fb);
+    // Gain node to compute the difference between the IIR and biquad filter.
+    var gain = context.createGain();
+    gain.gain.value = -1;
+
+    // Create the graph.
+    for (k = 0; k < nChannels; ++k)
+      source[k].connect(merger, 0, k);
+
+    merger.connect(biquad);
+    merger.connect(iir);
+    iir.connect(gain);
+    biquad.connect(context.destination);
+    gain.connect(context.destination);
+
+    for (k = 0; k < nChannels; ++k)
+      source[k].start();
+
+    return context.startRendering().then(function (result) {
+      var errorThresholds = [3.7671e-5, 3.0071e-5, 2.6241e-5];
+
+      // Check the difference signal on each channel
+      for (channel = 0; channel < result.numberOfChannels; ++channel) {
+        // Find the max amplitude of the result, which should be near zero.
+        var data = result.getChannelData(channel);
+        var maxError = data.reduce(function(reducedValue, currentValue) {
+          return Math.max(reducedValue, Math.abs(currentValue));
+        });
+
+        ok(maxError <= errorThresholds[channel], "Max difference between IIR and Biquad on channel " + channel);
+      }
+    });
+  }());
+
+  testPromises.push(function () {
+    // Apply an IIRFilter to the given input signal.
+    //
+    // IIR filter in the time domain is
+    //
+    //   y[n] = sum(ff[k]*x[n-k], k, 0, M) - sum(fb[k]*y[n-k], k, 1, N)
+    //
+    function iirFilter(input, feedforward, feedback) {
+      // For simplicity, create an x buffer that contains the input, and a y buffer that contains
+      // the output.  Both of these buffers have an initial work space to implement the initial
+      // memory of the filter.
+      var workSize = Math.max(feedforward.length, feedback.length);
+      var x = new Float32Array(input.length + workSize);
+
+      // Float64 because we want to match the implementation that uses doubles to minimize
+      // roundoff.
+      var y = new Float64Array(input.length + workSize);
+
+      // Copy the input over.
+      for (var k = 0; k < input.length; ++k)
+        x[k + feedforward.length] = input[k];
+
+      // Run the filter
+      for (var n = 0; n < input.length; ++n) {
+        var index = n + workSize;
+        var yn = 0;
+        for (var k = 0; k < feedforward.length; ++k)
+          yn += feedforward[k] * x[index - k];
+        for (var k = 0; k < feedback.length; ++k)
+          yn -= feedback[k] * y[index - k];
+
+        y[index] = yn;
+      }
+
+      return y.slice(workSize).map(Math.fround);
+    }
+
+    // Cascade the two given biquad filters to create one IIR filter.
+    function cascadeBiquads(f1Coef, f2Coef) {
+      // The biquad filters are:
+      //
+      // f1 = (b10 + b11/z + b12/z^2)/(1 + a11/z + a12/z^2);
+      // f2 = (b20 + b21/z + b22/z^2)/(1 + a21/z + a22/z^2);
+      //
+      // To cascade them, multiply the two transforms together to get a fourth order IIR filter.
+
+      var numProduct = [f1Coef.b0 * f2Coef.b0,
+        f1Coef.b0 * f2Coef.b1 + f1Coef.b1 * f2Coef.b0,
+        f1Coef.b0 * f2Coef.b2 + f1Coef.b1 * f2Coef.b1 + f1Coef.b2 * f2Coef.b0,
+        f1Coef.b1 * f2Coef.b2 + f1Coef.b2 * f2Coef.b1,
+        f1Coef.b2 * f2Coef.b2
+      ];
+
+      var denProduct = [1,
+        f2Coef.a1 + f1Coef.a1,
+        f2Coef.a2 + f1Coef.a1 * f2Coef.a1 + f1Coef.a2,
+        f1Coef.a1 * f2Coef.a2 + f1Coef.a2 * f2Coef.a1,
+        f1Coef.a2 * f2Coef.a2
+      ];
+
+      return {
+        ff: numProduct,
+        fb: denProduct
+      }
+    }
+
+    // Find the magnitude of the root of the quadratic that has the maximum magnitude.
+    //
+    // The quadratic is z^2 + a1 * z + a2 and we want the root z that has the largest magnitude.
+    function largestRootMagnitude(a1, a2) {
+      var discriminant = a1 * a1 - 4 * a2;
+      if (discriminant < 0) {
+        // Complex roots:  -a1/2 +/- i*sqrt(-d)/2.  Thus the magnitude of each root is the same
+        // and is sqrt(a1^2/4 + |d|/4)
+        var d = Math.sqrt(-discriminant);
+        return Math.hypot(a1 / 2, d / 2);
+      } else {
+        // Real roots
+        var d = Math.sqrt(discriminant);
+        return Math.max(Math.abs((-a1 + d) / 2), Math.abs((-a1 - d) / 2));
+      }
+    }
+
+    // Cascade 2 lowpass biquad filters and compare that with the equivalent 4th order IIR
+    // filter.
+
+    var nyquist = sampleRate / 2;
+    // Compute the coefficients of a lowpass filter.
+
+    // First some preliminary stuff.  Compute the coefficients of the biquad.  This is used to
+    // figure out how frames to use in the test.
+    var biquadType = "lowpass";
+    var biquadCutoff = 350;
+    var biquadQ = 5;
+    var biquadGain = 1;
+
+    var coef = createFilter(biquadType,
+      biquadCutoff / nyquist,
+      biquadQ,
+      biquadGain);
+
+    // Cascade the biquads together to create an equivalent IIR filter.
+    var cascade = cascadeBiquads(coef, coef);
+
+    // Since we're cascading two identical biquads, the root of denominator of the IIR filter is
+    // repeated, so the root of the denominator with the largest magnitude occurs twice.  The
+    // impulse response of the IIR filter will be roughly c*(r*r)^n at time n, where r is the
+    // root of largest magnitude.  This approximation gets better as n increases.  We can use
+    // this to get a rough idea of when the response has died down to a small value.
+
+    // This is the value we will use to determine how many frames to render.  Rendering too many
+    // is a waste of time and also makes it hard to compare the actual result to the expected
+    // because the magnitudes are so small that they could be mostly round-off noise.
+    //
+    // Find magnitude of the root with largest magnitude
+    var rootMagnitude = largestRootMagnitude(coef.a1, coef.a2);
+
+    // Find n such that |r|^(2*n) <= eps.  That is, n = log(eps)/(2*log(r)).  Somewhat
+    // arbitrarily choose eps = 1e-20;
+    var eps = 1e-20;
+    var framesForTest = Math.floor(Math.log(eps) / (2 * Math.log(rootMagnitude)));
+
+    // We're ready to create the graph for the test.  The offline context has two channels:
+    // channel 0 is the expected (cascaded biquad) result and channel 1 is the actual IIR filter
+    // result.
+    var context = new OfflineAudioContext(2, framesForTest, sampleRate);
+
+    // Use a simple impulse with a large (arbitrary) amplitude as the source
+    var amplitude = 1;
+    var buffer = context.createBuffer(1, testFrames, sampleRate);
+    buffer.getChannelData(0)[0] = amplitude;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Create the two biquad filters.  Doesn't really matter what, but for simplicity we choose
+    // identical lowpass filters with the same parameters.
+    var biquad1 = context.createBiquadFilter();
+    biquad1.type = biquadType;
+    biquad1.frequency.value = biquadCutoff;
+    biquad1.Q.value = biquadQ;
+
+    var biquad2 = context.createBiquadFilter();
+    biquad2.type = biquadType;
+    biquad2.frequency.value = biquadCutoff;
+    biquad2.Q.value = biquadQ;
+
+    var iir = context.createIIRFilter(cascade.ff, cascade.fb);
+
+    // Create the merger to get the signals into multiple channels
+    var merger = context.createChannelMerger(2);
+
+    // Create the graph, filtering the source through two biquads.
+    source.connect(biquad1);
+    biquad1.connect(biquad2);
+    biquad2.connect(merger, 0, 0);
+
+    source.connect(iir);
+    iir.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    // Now filter the source through the IIR filter.
+    var y = iirFilter(buffer.getChannelData(0), cascade.ff, cascade.fb);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function(result) {
+      var expected = result.getChannelData(0);
+      var actual = result.getChannelData(1);
+
+      compareChannels(actual, expected);
+
+    });
+  }());
+
+  // Wait for all tests
+  Promise.all(testPromises).then(function () {
+    SimpleTest.finish();
+  }, function () {
+    SimpleTest.finish();
+  });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
new file mode 100644
index 0000000000..cb5cf33ed6
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode GetFrequencyResponse</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <script type="text/javascript" src="biquad-filters.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  // Modified from WebKit/LayoutTests/webaudio/iirfilter-getFrequencyResponse.html
+  var sampleRate = 48000;
+  var testDurationSec = 0.01;
+
+  // Compute a set of linearly spaced frequencies.
+  function createFrequencies(nFrequencies, sampleRate)
+  {
+    var frequencies = new Float32Array(nFrequencies);
+    var nyquist = sampleRate / 2;
+    var freqDelta = nyquist / nFrequencies;
+
+    for (var k = 0; k < nFrequencies; ++k) {
+        frequencies[k] = k * freqDelta;
+    }
+
+    return frequencies;
+  }
+
+  // Number of frequency samples to take.
+  var numberOfFrequencies = 1000;
+
+  var context = new OfflineAudioContext(1, testDurationSec * sampleRate, sampleRate);
+
+  var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+
+  // 1-Pole IIR Filter
+  var iir = context.createIIRFilter([1], [1, -0.9]);
+
+  var iirMag = new Float32Array(numberOfFrequencies);
+  var iirPhase = new Float32Array(numberOfFrequencies);
+  var trueMag = new Float32Array(numberOfFrequencies);
+  var truePhase = new Float32Array(numberOfFrequencies);
+
+  // The IIR filter is
+  //   H(z) = 1/(1 - 0.9*z^(-1)).
+  //
+  // The frequency response is
+  //   H(exp(j*w)) = 1/(1 - 0.9*exp(-j*w)).
+  //
+  // Thus, the magnitude is
+  //   |H(exp(j*w))| = 1/sqrt(1.81-1.8*cos(w)).
+  //
+  // The phase is
+  //   arg(H(exp(j*w)) = atan(0.9*sin(w)/(.9*cos(w)-1))
+
+  var frequencyScale = Math.PI / (sampleRate / 2);
+
+  for (var k = 0; k < frequencies.length; ++k) {
+    var omega = frequencyScale * frequencies[k];
+    trueMag[k] = 1/Math.sqrt(1.81-1.8*Math.cos(omega));
+    truePhase[k] = Math.atan(0.9 * Math.sin(omega) / (0.9 * Math.cos(omega) - 1));
+  }
+
+  iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+  compareChannels(iirMag, trueMag);
+  compareChannels(iirPhase, truePhase);
+
+  // Compare IIR and Biquad Filter
+  // Create an IIR filter equivalent to the biquad filter. Compute the frequency response for both and verify that they are the same.
+  var biquad = context.createBiquadFilter();
+  var coef = createFilter(biquad.type,
+                          biquad.frequency.value / (context.sampleRate / 2),
+                          biquad.Q.value,
+                          biquad.gain.value);
+
+  var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+  var biquadMag = new Float32Array(numberOfFrequencies);
+  var biquadPhase = new Float32Array(numberOfFrequencies);
+  var iirMag = new Float32Array(numberOfFrequencies);
+  var iirPhase = new Float32Array(numberOfFrequencies);
+
+  biquad.getFrequencyResponse(frequencies, biquadMag, biquadPhase);
+  iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+  compareChannels(iirMag, biquadMag);
+  compareChannels(iirPhase, biquadPhase);
+
+  SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/browser.ini b/dom/media/webaudio/test/browser.ini
new file mode 100644
index 0000000000..60ed969f1a
--- /dev/null
+++ b/dom/media/webaudio/test/browser.ini
@@ -0,0 +1 @@
+[browser_bug1181073.js]
\ No newline at end of file
diff --git a/dom/media/webaudio/test/browser_bug1181073.js b/dom/media/webaudio/test/browser_bug1181073.js
new file mode 100644
index 0000000000..6ee48144c4
--- /dev/null
+++ b/dom/media/webaudio/test/browser_bug1181073.js
@@ -0,0 +1,40 @@
+add_task(function*() {
+  // Make the min_background_timeout_value very high to avoid problems on slow machines
+  yield new Promise(resolve => SpecialPowers.pushPrefEnv({
+    'set': [['dom.min_background_timeout_value', 3000]]
+  }, resolve));
+
+  // Make a new tab, and put it in the background
+  yield BrowserTestUtils.withNewTab("about:blank", function*(browser) {
+    yield BrowserTestUtils.withNewTab("about:blank", function*() {
+      let time = yield ContentTask.spawn(browser, null, function () {
+        return new Promise(resolve => {
+          let start = content.performance.now();
+          let id = content.window.setInterval(function() {
+            let end = content.performance.now();
+            content.window.clearInterval(id);
+            resolve(end - start);
+          }, 0);
+        });
+      });
+
+      ok(time > 2000, "Interval is throttled with no webaudio (" + time + " ms)");
+
+      time = yield ContentTask.spawn(browser, null, function () {
+        return new Promise(resolve => {
+          // Create an audio context, and save it on the window so it doesn't get GCed
+          content.window._audioCtx = new content.window.AudioContext();
+
+          let start = content.performance.now();
+          let id = content.window.setInterval(function() {
+            let end = content.performance.now();
+            content.window.clearInterval(id);
+            resolve(end - start);
+          }, 0);
+        });
+      });
+
+      ok(time < 1000, "Interval is not throttled with an audio context present (" + time + " ms)");
+    });
+  });
+});
diff --git a/dom/media/webaudio/test/corsServer.sjs b/dom/media/webaudio/test/corsServer.sjs
new file mode 100644
index 0000000000..1804c7862f
--- /dev/null
+++ b/dom/media/webaudio/test/corsServer.sjs
@@ -0,0 +1,25 @@
+function handleRequest(request, response)
+{
+  var file = Components.classes["@mozilla.org/file/directory_service;1"].
+                        getService(Components.interfaces.nsIProperties).
+                        get("CurWorkD", Components.interfaces.nsILocalFile);
+  var fis  = Components.classes['@mozilla.org/network/file-input-stream;1'].
+                        createInstance(Components.interfaces.nsIFileInputStream);
+  var bis  = Components.classes["@mozilla.org/binaryinputstream;1"].
+                        createInstance(Components.interfaces.nsIBinaryInputStream);
+  var paths = "tests/dom/media/webaudio/test/small-shot.ogg";
+  var split = paths.split("/");
+  for(var i = 0; i < split.length; ++i) {
+    file.append(split[i]);
+  }
+  fis.init(file, -1, -1, false);
+  bis.setInputStream(fis);
+  var bytes = bis.readBytes(bis.available());
+  response.setHeader("Content-Type", "video/ogg", false);
+  response.setHeader("Content-Length", ""+ bytes.length, false);
+  response.setHeader("Access-Control-Allow-Origin", "*", false);
+  response.write(bytes, bytes.length);
+  response.processAsync();
+  response.finish();
+  bis.close();
+}
diff --git a/dom/media/webaudio/test/invalid.txt b/dom/media/webaudio/test/invalid.txt
new file mode 100644
index 0000000000..c44840faf1
--- /dev/null
+++ b/dom/media/webaudio/test/invalid.txt
@@ -0,0 +1 @@
+Surely this is not an audio file!
diff --git a/dom/media/webaudio/test/layouttest-glue.js b/dom/media/webaudio/test/layouttest-glue.js
new file mode 100644
index 0000000000..db1aa563b4
--- /dev/null
+++ b/dom/media/webaudio/test/layouttest-glue.js
@@ -0,0 +1,19 @@
+// Reimplementation of the LayoutTest API from Blink so we can easily port
+// WebAudio tests to Simpletest, without touching the internals of the test.
+
+function testFailed(msg) {
+  ok(false, msg);
+}
+
+function testPassed(msg) {
+  ok(true, msg);
+}
+
+function finishJSTest() {
+  SimpleTest.finish();
+}
+
+function description(str) {
+  info(str);
+}
+
diff --git a/dom/media/webaudio/test/mochitest.ini b/dom/media/webaudio/test/mochitest.ini
new file mode 100644
index 0000000000..4abcce7e30
--- /dev/null
+++ b/dom/media/webaudio/test/mochitest.ini
@@ -0,0 +1,212 @@
+[DEFAULT]
+tags=msg
+tags = webaudio
+subsuite = media
+support-files =
+  audio-expected.wav
+  audio-mono-expected-2.wav
+  audio-mono-expected.wav
+  audio-quad.wav
+  audio.ogv
+  audiovideo.mp4
+  audioBufferSourceNodeDetached_worker.js
+  corsServer.sjs
+  invalid.txt
+  layouttest-glue.js
+  noaudio.webm
+  small-shot-expected.wav
+  small-shot-mono-expected.wav
+  small-shot.ogg
+  small-shot.mp3
+  sweep-300-330-1sec.opus
+  ting-44.1k-1ch.ogg
+  ting-44.1k-2ch.ogg
+  ting-48k-1ch.ogg
+  ting-48k-2ch.ogg
+  ting-44.1k-1ch.wav
+  ting-44.1k-2ch.wav
+  ting-48k-1ch.wav
+  ting-48k-2ch.wav
+  sine-440-10s.opus
+  webaudio.js
+
+[test_analyserNode.html]
+[test_analyserScale.html]
+[test_analyserNodeOutput.html]
+[test_analyserNodePassThrough.html]
+[test_analyserNodeWithGain.html]
+[test_AudioBuffer.html]
+[test_audioBufferSourceNode.html]
+[test_audioBufferSourceNodeEnded.html]
+[test_audioBufferSourceNodeLazyLoopParam.html]
+[test_audioBufferSourceNodeLoop.html]
+[test_audioBufferSourceNodeLoopStartEnd.html]
+[test_audioBufferSourceNodeLoopStartEndSame.html]
+[test_audioBufferSourceNodeDetached.html]
+skip-if = (toolkit == 'android' && debug) || os == 'win' # bug 1127845, bug 1138468
+[test_audioBufferSourceNodeNoStart.html]
+[test_audioBufferSourceNodeNullBuffer.html]
+[test_audioBufferSourceNodeOffset.html]
+skip-if = (toolkit == 'android') || debug #bug 906752
+[test_audioBufferSourceNodePassThrough.html]
+[test_audioBufferSourceNodeRate.html]
+[test_AudioContext.html]
+[test_AudioContext_disabled.html]
+[test_audioContextSuspendResumeClose.html]
+tags=capturestream
+[test_audioDestinationNode.html]
+[test_AudioListener.html]
+[test_AudioNodeDevtoolsAPI.html]
+[test_audioParamChaining.html]
+[test_AudioParamDevtoolsAPI.html]
+[test_audioParamExponentialRamp.html]
+[test_audioParamGain.html]
+[test_audioParamLinearRamp.html]
+[test_audioParamSetCurveAtTime.html]
+[test_audioParamSetCurveAtTimeTwice.html]
+[test_audioParamSetCurveAtTimeZeroDuration.html]
+[test_audioParamSetTargetAtTime.html]
+[test_audioParamSetTargetAtTimeZeroTimeConstant.html]
+[test_audioParamSetValueAtTime.html]
+[test_audioParamTimelineDestinationOffset.html]
+[test_badConnect.html]
+[test_biquadFilterNode.html]
+[test_biquadFilterNodePassThrough.html]
+[test_biquadFilterNodeWithGain.html]
+[test_bug808374.html]
+[test_bug827541.html]
+[test_bug839753.html]
+[test_bug845960.html]
+[test_bug856771.html]
+[test_bug866570.html]
+[test_bug866737.html]
+[test_bug867089.html]
+[test_bug867104.html]
+[test_bug867174.html]
+[test_bug867203.html]
+[test_bug875221.html]
+[test_bug875402.html]
+[test_bug894150.html]
+[test_bug956489.html]
+[test_bug964376.html]
+[test_bug966247.html]
+tags=capturestream
+[test_bug972678.html]
+[test_bug1113634.html]
+[test_bug1118372.html]
+[test_bug1027864.html]
+[test_bug1056032.html]
+skip-if = toolkit == 'android' # bug 1056706
+[test_bug1255618.html]
+[test_bug1267579.html]
+[test_channelMergerNode.html]
+[test_channelMergerNodeWithVolume.html]
+[test_channelSplitterNode.html]
+[test_channelSplitterNodeWithVolume.html]
+skip-if = (android_version == '18' && debug) # bug 1158417
+[test_convolverNode.html]
+[test_convolverNode_mono_mono.html]
+[test_convolverNodeChannelCount.html]
+[test_convolverNodeDelay.html]
+[test_convolverNodeFiniteInfluence.html]
+[test_convolverNodePassThrough.html]
+[test_convolverNodeWithGain.html]
+[test_currentTime.html]
+[test_decodeMultichannel.html]
+[test_decodeAudioDataPromise.html]
+[test_decodeOpusTail.html]
+[test_delayNode.html]
+[test_delayNodeAtMax.html]
+[test_delayNodeChannelChanges.html]
+skip-if = toolkit == 'android' # bug 1056706
+[test_delayNodeCycles.html]
+[test_delayNodePassThrough.html]
+[test_delayNodeSmallMaxDelay.html]
+[test_delayNodeTailIncrease.html]
+[test_delayNodeTailWithDisconnect.html]
+[test_delayNodeTailWithGain.html]
+[test_delayNodeTailWithReconnect.html]
+[test_delayNodeWithGain.html]
+[test_disconnectAll.html]
+[test_disconnectAudioParam.html]
+[test_disconnectAudioParamFromOutput.html]
+[test_disconnectExceptions.html]
+[test_disconnectFromAudioNode.html]
+[test_disconnectFromAudioNodeAndOutput.html]
+[test_disconnectFromAudioNodeAndOutputAndInput.html]
+[test_disconnectFromAudioNodeMultipleConnection.html]
+[test_disconnectFromOutput.html]
+[test_dynamicsCompressorNode.html]
+[test_dynamicsCompressorNodePassThrough.html]
+[test_dynamicsCompressorNodeWithGain.html]
+[test_gainNode.html]
+[test_gainNodeInLoop.html]
+[test_gainNodePassThrough.html]
+[test_iirFilterNodePassThrough.html]
+[test_maxChannelCount.html]
+[test_mediaDecoding.html]
+[test_mediaElementAudioSourceNode.html]
+tags=capturestream
+[test_mediaElementAudioSourceNodeFidelity.html]
+tags=capturestream
+[test_mediaElementAudioSourceNodePassThrough.html]
+tags=capturestream
+skip-if = toolkit == 'android' # bug 1145816
+[test_mediaElementAudioSourceNodeVideo.html]
+tags=capturestream
+[test_mediaElementAudioSourceNodeCrossOrigin.html]
+tags=capturestream
+skip-if = toolkit == 'android' # bug 1145816
+[test_mediaStreamAudioDestinationNode.html]
+[test_mediaStreamAudioSourceNode.html]
+[test_mediaStreamAudioSourceNodeCrossOrigin.html]
+tags=capturestream
+[test_mediaStreamAudioSourceNodeNoGC.html]
+[test_mediaStreamAudioSourceNodePassThrough.html]
+[test_mediaStreamAudioSourceNodeResampling.html]
+tags=capturestream
+[test_mixingRules.html]
+skip-if = toolkit == 'android' # bug 1091965
+[test_mozaudiochannel.html]
+# Android: bug 1061675; OSX 10.6: bug 1097721
+skip-if = (toolkit == 'android') || (os == 'mac' && os_version == '10.6')
+[test_nodeToParamConnection.html]
+[test_OfflineAudioContext.html]
+[test_offlineDestinationChannelCountLess.html]
+[test_offlineDestinationChannelCountMore.html]
+[test_oscillatorNode.html]
+[test_oscillatorNode2.html]
+[test_oscillatorNodeNegativeFrequency.html]
+[test_oscillatorNodePassThrough.html]
+[test_oscillatorNodeStart.html]
+[test_oscillatorTypeChange.html]
+[test_pannerNode.html]
+[test_pannerNode_equalPower.html]
+[test_pannerNodeAbove.html]
+[test_pannerNodeAtZeroDistance.html]
+[test_pannerNodeChannelCount.html]
+[test_pannerNodeHRTFSymmetry.html]
+[test_pannerNodeTail.html]
+[test_pannerNode_maxDistance.html]
+[test_stereoPannerNode.html]
+[test_stereoPannerNodePassThrough.html]
+[test_periodicWave.html]
+[test_periodicWaveDisableNormalization.html]
+[test_periodicWaveBandLimiting.html]
+[test_ScriptProcessorCollected1.html]
+[test_scriptProcessorNode.html]
+[test_scriptProcessorNodeChannelCount.html]
+[test_scriptProcessorNodePassThrough.html]
+[test_scriptProcessorNode_playbackTime1.html]
+[test_scriptProcessorNodeZeroInputOutput.html]
+[test_scriptProcessorNodeNotConnected.html]
+[test_sequentialBufferSourceWithResampling.html]
+[test_singleSourceDest.html]
+[test_stereoPanningWithGain.html]
+[test_waveDecoder.html]
+[test_waveShaper.html]
+[test_waveShaperGain.html]
+[test_waveShaperNoCurve.html]
+[test_waveShaperPassThrough.html]
+[test_waveShaperInvalidLengthCurve.html]
+[test_WebAudioMemoryReporting.html]
diff --git a/dom/media/webaudio/test/noaudio.webm b/dom/media/webaudio/test/noaudio.webm
new file mode 100644
index 0000000000..9207017fb6
--- /dev/null
+++ b/dom/media/webaudio/test/noaudio.webm
diff --git a/dom/media/webaudio/test/sine-440-10s.opus b/dom/media/webaudio/test/sine-440-10s.opus
new file mode 100644
index 0000000000..eb91020168
--- /dev/null
+++ b/dom/media/webaudio/test/sine-440-10s.opus
diff --git a/dom/media/webaudio/test/small-shot-expected.wav b/dom/media/webaudio/test/small-shot-expected.wav
new file mode 100644
index 0000000000..2faaa8258b
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot-expected.wav
diff --git a/dom/media/webaudio/test/small-shot-mono-expected.wav b/dom/media/webaudio/test/small-shot-mono-expected.wav
new file mode 100644
index 0000000000..d4e2647e42
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot-mono-expected.wav
diff --git a/dom/media/webaudio/test/small-shot.mp3 b/dom/media/webaudio/test/small-shot.mp3
new file mode 100644
index 0000000000..f9397a5106
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot.mp3
diff --git a/dom/media/webaudio/test/small-shot.ogg b/dom/media/webaudio/test/small-shot.ogg
new file mode 100644
index 0000000000..1a41623f81
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot.ogg
diff --git a/dom/media/webaudio/test/sweep-300-330-1sec.opus b/dom/media/webaudio/test/sweep-300-330-1sec.opus
new file mode 100644
index 0000000000..619d1e0844
--- /dev/null
+++ b/dom/media/webaudio/test/sweep-300-330-1sec.opus
diff --git a/dom/media/webaudio/test/test_AudioBuffer.html b/dom/media/webaudio/test/test_AudioBuffer.html
new file mode 100644
index 0000000000..82bfdd420a
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioBuffer.html
@@ -0,0 +1,105 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  SpecialPowers.gc(); // Make sure that our channels are accessible after GC
+  ok(buffer, "Buffer was allocated successfully");
+  is(buffer.sampleRate, context.sampleRate, "Correct sample rate");
+  is(buffer.length, 2048, "Correct length");
+  ok(Math.abs(buffer.duration - 2048 / context.sampleRate) < 0.0001, "Correct duration");
+  is(buffer.numberOfChannels, 2, "Correct number of channels");
+  for (var i = 0; i < buffer.numberOfChannels; ++i) {
+    var buf = buffer.getChannelData(i);
+    ok(buf, "Buffer index " + i + " exists");
+    ok(buf instanceof Float32Array, "Result is a typed array");
+    is(buf.length, buffer.length, "Correct length");
+    var foundNonZero = false;
+    for (var j = 0; j < buf.length; ++j) {
+      if (buf[j] != 0) {
+        foundNonZero = true;
+        break;
+      }
+      buf[j] = j;
+    }
+    ok(!foundNonZero, "Buffer " + i + " should be initialized to 0");
+  }
+
+  // Now test copying the channel data out of a normal buffer
+  var copy = new Float32Array(100);
+  buffer.copyFromChannel(copy, 0, 1024);
+  for (var i = 0; i < copy.length; ++i) {
+    is(copy[i], 1024 + i, "Correct sample");
+  }
+
+  // Test copying the channel data out of a playing buffer
+  var srcNode = context.createBufferSource();
+  srcNode.buffer = buffer;
+  srcNode.start(0);
+  copy = new Float32Array(100);
+  buffer.copyFromChannel(copy, 0, 1024);
+  for (var i = 0; i < copy.length; ++i) {
+    is(copy[i], 1024 + i, "Correct sample");
+  }
+
+  // Test copying to the channel data
+  var newData = new Float32Array(200);
+  buffer.copyToChannel(newData, 0, 100);
+  var changedData = buffer.getChannelData(0);
+  for (var i = 0; i < changedData.length; ++i) {
+    if (i < 100 || i >= 300) {
+      is(changedData[i], i, "Untouched sample");
+    } else {
+      is(changedData[i], 0, "Correct sample");
+    }
+  }
+
+  // Now, detach the array buffer
+  var worker = new Worker("audioBufferSourceNodeDetached_worker.js");
+  var data = buffer.getChannelData(0).buffer;
+  worker.postMessage(data, [data]);
+  SpecialPowers.gc();
+
+  expectException(function() {
+    buffer.copyFromChannel(copy, 0, 1024);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  expectException(function() {
+    buffer.copyToChannel(newData, 0, 100);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  expectException(function() {
+    context.createBuffer(2, 2048, 7999);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createBuffer(2, 2048, 192001);
+  }, DOMException.INDEX_SIZE_ERR);
+  context.createBuffer(2, 2048, 8000);  // no exception
+  context.createBuffer(2, 2048, 192000); // no exception
+  context.createBuffer(32, 2048, 48000); // no exception
+  // Null length
+  expectException(function() {
+    context.createBuffer(2, 0, 48000);
+  }, DOMException.INDEX_SIZE_ERR);
+  // Null number of channels
+  expectException(function() {
+    context.createBuffer(0, 2048, 48000);
+  }, DOMException.INDEX_SIZE_ERR);
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioContext.html b/dom/media/webaudio/test/test_AudioContext.html
new file mode 100644
index 0000000000..0cab4354e8
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioContext.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ok(ac, "Create a AudioContext object");
+  ok(ac instanceof EventTarget, "AudioContexts must be EventTargets");
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioContext_disabled.html b/dom/media/webaudio/test/test_AudioContext_disabled.html
new file mode 100644
index 0000000000..03e0775edd
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioContext_disabled.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can disable the AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+const webaudio_interfaces = [
+  "AudioContext",
+  "OfflineAudioContext",
+  "AudioContext",
+  "OfflineAudioCompletionEvent",
+  "AudioNode",
+  "AudioDestinationNode",
+  "AudioParam",
+  "GainNode",
+  "DelayNode",
+  "AudioBuffer",
+  "AudioBufferSourceNode",
+  "MediaElementAudioSourceNode",
+  "ScriptProcessorNode",
+  "AudioProcessingEvent",
+  "PannerNode",
+  "AudioListener",
+  "StereoPannerNode",
+  "ConvolverNode",
+  "AnalyserNode",
+  "ChannelSplitterNode",
+  "ChannelMergerNode",
+  "DynamicsCompressorNode",
+  "BiquadFilterNode",
+  "IIRFilterNode",
+  "WaveShaperNode",
+  "OscillatorNode",
+  "PeriodicWave",
+  "MediaStreamAudioSourceNode",
+  "MediaStreamAudioDestinationNode"
+];
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  SpecialPowers.pushPrefEnv({"set": [["dom.webaudio.enabled", false]]}, function() {
+      webaudio_interfaces.forEach((e) => ok(!window[e], e + " must be disabled when the Web Audio API is disabled"));
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioListener.html b/dom/media/webaudio/test/test_AudioListener.html
new file mode 100644
index 0000000000..07ad154d76
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioListener.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioContext.listener and the AudioListener interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  ok("listener" in context, "AudioContext.listener should exist");
+  ok(Math.abs(context.listener.dopplerFactor - 1.0) < 1e-4, "Correct default doppler factor");
+  ok(Math.abs(context.listener.speedOfSound - 343.3) < 1e-4, "Correct default speed of sound value");
+  context.listener.dopplerFactor = 0.5;
+  ok(Math.abs(context.listener.dopplerFactor - 0.5) < 1e-4, "The doppler factor value can be changed");
+  context.listener.speedOfSound = 400;
+  ok(Math.abs(context.listener.speedOfSound - 400) < 1e-4, "The speed of sound can be changed");
+  // The values set by the following cannot be read from script, but the
+  // implementation is simple enough, so we just make sure that nothing throws.
+  with (context.listener) {
+    setPosition(1.0, 1.0, 1.0);
+    setOrientation(1.0, 1.0, 1.0, 1.0, 1.0, 1.0);
+    setVelocity(0.5, 1.0, 1.5);
+  }
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html b/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html
new file mode 100644
index 0000000000..49f71505d1
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the devtool AudioNode API</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+  SimpleTest.waitForExplicitFinish();
+
+  function id(node) {
+    return SpecialPowers.getPrivilegedProps(node, "id");
+  }
+
+  var ac = new AudioContext();
+  var ids;
+  var weak;
+  (function() {
+    var src1 = ac.createBufferSource();
+    var src2 = ac.createBufferSource();
+    ok(id(src2) > id(src1), "The ID should be monotonic");
+    ok(id(src1) > id(ac.destination), "The ID of the destination node should be the lowest");
+    ids = [id(src1), id(src2)];
+    weak = SpecialPowers.Cu.getWeakReference(src1);
+    is(SpecialPowers.unwrap(weak.get()), src1, "The node should support a weak reference");
+  })();
+  function observer(subject, topic, data) {
+    var id = parseInt(data);
+    var index = ids.indexOf(id);
+    if (index != -1) {
+      info("Dropping id " + id + " at index " + index);
+      ids.splice(index, 1);
+      if (ids.length == 0) {
+        SimpleTest.executeSoon(function() {
+          is(weak.get(), null, "The weak reference must be dropped now");
+          SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+          SimpleTest.finish();
+        });
+      }
+    }
+  }
+  SpecialPowers.addObserver(observer, "webaudio-node-demise", false);
+
+  forceCC();
+  forceCC();
+
+  function forceCC() {
+    SpecialPowers.DOMWindowUtils.cycleCollect();
+    SpecialPowers.DOMWindowUtils.garbageCollect();
+    SpecialPowers.DOMWindowUtils.garbageCollect();
+  }
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html b/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html
new file mode 100644
index 0000000000..9b59dda8a9
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the devtool AudioParam API</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+  function checkIdAndName(node, name) {
+    is(SpecialPowers.getPrivilegedProps(node, "id"),
+       SpecialPowers.getPrivilegedProps(node[name], "parentNodeId"),
+       "The parent id should be correct");
+    is(SpecialPowers.getPrivilegedProps(node[name], "name"), name,
+       "The name of the AudioParam should be correct.");
+  }
+
+  var ac = new AudioContext(),
+      gain = ac.createGain(),
+      osc = ac.createOscillator(),
+      del = ac.createDelay(),
+      source = ac.createBufferSource(),
+      stereoPanner = ac.createStereoPanner(),
+      comp = ac.createDynamicsCompressor(),
+      biquad = ac.createBiquadFilter();
+
+  checkIdAndName(gain, "gain");
+  checkIdAndName(osc, "frequency");
+  checkIdAndName(osc, "detune");
+  checkIdAndName(del, "delayTime");
+  checkIdAndName(source, "playbackRate");
+  checkIdAndName(source, "detune");
+  checkIdAndName(stereoPanner, "pan");
+  checkIdAndName(comp, "threshold");
+  checkIdAndName(comp, "knee");
+  checkIdAndName(comp, "ratio");
+  checkIdAndName(comp, "attack");
+  checkIdAndName(comp, "release");
+  checkIdAndName(biquad, "frequency");
+  checkIdAndName(biquad, "detune");
+  checkIdAndName(biquad, "Q");
+  checkIdAndName(biquad, "gain");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_OfflineAudioContext.html b/dom/media/webaudio/test/test_OfflineAudioContext.html
new file mode 100644
index 0000000000..81d3e2313c
--- /dev/null
+++ b/dom/media/webaudio/test/test_OfflineAudioContext.html
@@ -0,0 +1,102 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var renderedBuffer = null;
+var finished = 0;
+
+function finish() {
+  finished++;
+  if (finished == 2) {
+    SimpleTest.finish();
+  }
+}
+
+function setOrCompareRenderedBuffer(aRenderedBuffer) {
+  if (renderedBuffer) {
+    is(renderedBuffer, aRenderedBuffer, "Rendered buffers from the event and the promise should be the same");
+    finish();
+  } else {
+    renderedBuffer = aRenderedBuffer;
+  }
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new OfflineAudioContext(2, 100, 22050);
+  ok(ctx instanceof EventTarget, "OfflineAudioContexts must be EventTargets");
+  is(ctx.length, 100, "OfflineAudioContext.length is equal to the value passed to the ctor.");
+
+  var buf = ctx.createBuffer(2, 100, ctx.sampleRate);
+  for (var i = 0; i < 2; ++i) {
+    for (var j = 0; j < 100; ++j) {
+      buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+    }
+  }
+
+  expectException(function() {
+    ctx.createMediaStreamDestination();
+  }, DOMException.NOT_SUPPORTED_ERR);
+
+  expectException(function() {
+    new OfflineAudioContext(2, 100, 0);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(2, 100, -1);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(0, 100, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  new OfflineAudioContext(32, 100, 44100);
+  expectException(function() {
+    new OfflineAudioContext(33, 100, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(2, 0, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+
+  ctx.addEventListener("complete", function(e) {
+    ok(e instanceof OfflineAudioCompletionEvent, "Correct event received");
+    is(e.renderedBuffer.numberOfChannels, 2, "Correct expected number of buffers");
+    ok(renderedBuffer != null, "The event should be fired after the promise callback.");
+    expectNoException(function() {
+      ctx.startRendering().then(function() {
+        ok(false, "Promise should not resolve when startRendering is called a second time on an OfflineAudioContext")
+        finish();
+      }).catch(function(err) {
+        ok(true, "Promise should reject when startRendering is called a second time on an OfflineAudioContext")
+        finish();
+      });
+    });
+    compareBuffers(e.renderedBuffer, buf);
+    setOrCompareRenderedBuffer(e.renderedBuffer);
+
+  }, false);
+
+  expectNoException(function() {
+    ctx.startRendering().then(function(b) {
+      is(renderedBuffer, null, "The promise callback should be called first.");
+      setOrCompareRenderedBuffer(b);
+    }).catch(function (error) {
+       ok(false, "The promise from OfflineAudioContext.startRendering should never be rejected");
+    });
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_ScriptProcessorCollected1.html b/dom/media/webaudio/test/test_ScriptProcessorCollected1.html
new file mode 100644
index 0000000000..931f995dfc
--- /dev/null
+++ b/dom/media/webaudio/test/test_ScriptProcessorCollected1.html
@@ -0,0 +1,84 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode in cycle with no listener is collected</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var observer = function(subject, topic, data) {
+  var id = parseInt(data);
+  var index = ids.indexOf(id);
+  if (index != -1) {
+    ok(true, "Collected AudioNode id " + id + " at index " + index);
+    ids.splice(index, 1);
+  }
+}
+
+SpecialPowers.addObserver(observer, "webaudio-node-demise", false);
+
+SimpleTest.registerCleanupFunction(function() {
+  if (observer) {
+    SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+  }
+});
+
+var ac = new AudioContext();
+
+var testProcessor = ac.createScriptProcessor(256, 1, 0);
+var delay = ac.createDelay();
+testProcessor.connect(delay);
+delay.connect(testProcessor);
+
+var referenceProcessor = ac.createScriptProcessor(256, 1, 0);
+var gain = ac.createGain();
+gain.connect(referenceProcessor);
+
+var processCount = 0;
+testProcessor.onaudioprocess = function(event) {
+  ++processCount;
+  switch (processCount) {
+  case 1:
+    // Switch to listening to referenceProcessor;
+    referenceProcessor.onaudioprocess = event.target.onaudioprocess;
+    referenceProcessor = null;
+    event.target.onaudioprocess = null;
+  case 2:
+    // There are no references to testProcessor and so GC can begin.
+    SpecialPowers.forceGC();
+    break;
+  case 3:
+    // Another GC should not be required after testProcessor would have
+    // received another audioprocess event.
+    SpecialPowers.forceCC();
+    // Expect that webaudio-demise has been queued.
+    // Queue another event to check.
+    SimpleTest.executeSoon(function() {
+      SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+      observer = null;
+      event.target.onaudioprocess = null;
+      ok(ids.length == 0, "All expected nodes should be collected");
+      SimpleTest.finish();
+    });
+    break;
+  }
+};
+
+function id(node) {
+  return SpecialPowers.getPrivilegedProps(node, "id");
+}
+
+// Nodes with these ids should be collected.
+var ids = [ id(testProcessor), id(delay), id(gain) ];
+testProcessor = null;
+delay = null;
+gain = null;
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_WebAudioMemoryReporting.html b/dom/media/webaudio/test/test_WebAudioMemoryReporting.html
new file mode 100644
index 0000000000..c753756e79
--- /dev/null
+++ b/dom/media/webaudio/test/test_WebAudioMemoryReporting.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Web Audio memory reporting</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ac = new AudioContext();
+var sp = ac.createScriptProcessor(4096, 1, 1);
+sp.connect(ac.destination);
+
+// Not started so as to test
+// https://bugzilla.mozilla.org/show_bug.cgi?id=1225003#c2
+var oac = new OfflineAudioContext(1, 1, 48000);
+
+var nodeTypes = ["ScriptProcessorNode", "AudioDestinationNode"];
+var objectTypes = ["dom-nodes", "engine-objects", "stream-objects"];
+
+var usages = { "explicit/webaudio/audiocontext": 0 };
+
+for (var i = 0; i < nodeTypes.length; ++i) {
+  for (var j = 0; j < objectTypes.length; ++j) {
+    usages["explicit/webaudio/audio-node/" +
+           nodeTypes[i] + "/" + objectTypes[j]] = 0;
+  }
+}
+
+var handleReport = function(aProcess, aPath, aKind, aUnits, aAmount, aDesc) {
+  if (aPath in usages) {
+    usages[aPath] += aAmount;
+  }
+}
+
+var finished = function () {
+  ok(true, "Yay didn't crash!");
+  for (var resource in usages) {
+    ok(usages[resource] > 0, "Non-zero usage for " + resource);
+  };
+  SimpleTest.finish();
+}
+
+SpecialPowers.Cc["@mozilla.org/memory-reporter-manager;1"].
+  getService(SpecialPowers.Ci.nsIMemoryReporterManager).
+  getReports(handleReport, null, finished, null, /* anonymized = */ false);
+
+// To test bug 1225003, run a failing decodeAudioData() job over a time when
+// the tasks from getReports() are expected to run.
+ac.decodeAudioData(new ArrayBuffer(4), function(){}, function(){});
+</script>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNode.html b/dom/media/webaudio/test/test_analyserNode.html
new file mode 100644
index 0000000000..7af67a5a56
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNode.html
@@ -0,0 +1,103 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var analyser = context.createAnalyser();
+
+  source.buffer = buffer;
+
+  source.connect(analyser);
+  analyser.connect(destination);
+
+  is(analyser.channelCount, 1, "analyser node has 1 input channels by default");
+  is(analyser.channelCountMode, "max", "Correct channelCountMode for the analyser node");
+  is(analyser.channelInterpretation, "speakers", "Correct channelCountInterpretation for the analyser node");
+
+  is(analyser.fftSize, 2048, "Correct default value for fftSize");
+  is(analyser.frequencyBinCount, 1024, "Correct default value for frequencyBinCount");
+  expectException(function() {
+    analyser.fftSize = 0;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 1;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 8;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 100; // non-power of two
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 2049;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 4097;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 8193;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 16385;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 32769;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 65536;
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser.fftSize = 1024;
+  is(analyser.frequencyBinCount, 512, "Correct new value for frequencyBinCount");
+
+  is(analyser.minDecibels, -100, "Correct default value for minDecibels");
+  is(analyser.maxDecibels, -30, "Correct default value for maxDecibels");
+  expectException(function() {
+    analyser.minDecibels = -30;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.minDecibels = -29;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.maxDecibels = -100;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.maxDecibels = -101;
+  }, DOMException.INDEX_SIZE_ERR);
+
+  is(analyser.smoothingTimeConstant, 0.8, "Correct default value for smoothingTimeConstant");
+  expectException(function() {
+    analyser.smoothingTimeConstant = -0.1;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.smoothingTimeConstant = 1.1;
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser.smoothingTimeConstant = 0;
+  analyser.smoothingTimeConstant = 1;
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodeOutput.html b/dom/media/webaudio/test/test_analyserNodeOutput.html
new file mode 100644
index 0000000000..e6255fee00
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodeOutput.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var analyser = context.createAnalyser();
+
+    source.buffer = this.buffer;
+
+    source.connect(analyser);
+
+    source.start(0);
+    return analyser;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodePassThrough.html b/dom/media/webaudio/test/test_analyserNodePassThrough.html
new file mode 100644
index 0000000000..37d1db5103
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var analyser = context.createAnalyser();
+
+    source.buffer = this.buffer;
+
+    source.connect(analyser);
+
+    var analyserWrapped = SpecialPowers.wrap(analyser);
+    ok("passThrough" in analyserWrapped, "AnalyserNode should support the passThrough API");
+    analyserWrapped.passThrough = true;
+
+    source.start(0);
+    return analyser;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodeWithGain.html b/dom/media/webaudio/test/test_analyserNodeWithGain.html
new file mode 100644
index 0000000000..fa0a2caa75
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodeWithGain.html
@@ -0,0 +1,47 @@
+<!DOCTYPE html>
+<title>Test effect of AnalyserNode on GainNode output</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+promise_test(function() {
+  // fftSize <= bufferSize so that the time domain data is full of input after
+  // processing the buffer.
+  const fftSize = 32;
+  const bufferSize = 128;
+
+  var context = new OfflineAudioContext(1, bufferSize, 48000);
+
+  var analyser1 = context.createAnalyser();
+  analyser1.fftSize = fftSize;
+  analyser1.connect(context.destination);
+  var analyser2 = context.createAnalyser();
+  analyser2.fftSize = fftSize;
+
+  var gain = context.createGain();
+  gain.gain.value = 2.0;
+  gain.connect(analyser1);
+  gain.connect(analyser2);
+
+  // Create a DC input to make getFloatTimeDomainData() output consistent at
+  // any time.
+  var buffer = context.createBuffer(1, 1, context.sampleRate);
+  buffer.getChannelData(0)[0] = 1.0 / gain.gain.value;
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+  source.loop = true;
+  source.connect(gain);
+  source.start();
+
+  return context.startRendering().
+    then(function(buffer) {
+      assert_equals(buffer.getChannelData(0)[0], 1.0,
+                    "analyser1 output");
+
+      var data = new Float32Array(1);
+      analyser1.getFloatTimeDomainData(data);
+      assert_equals(data[0], 1.0, "analyser1 time domain data");
+      analyser2.getFloatTimeDomainData(data);
+      assert_equals(data[0], 1.0, "analyser2 time domain data");
+    });
+});
+</script>
diff --git a/dom/media/webaudio/test/test_analyserScale.html b/dom/media/webaudio/test/test_analyserScale.html
new file mode 100644
index 0000000000..3aec8d22b2
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserScale.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode when the input is scaled </title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+
+  var gain = context.createGain();
+  var analyser = context.createAnalyser();
+  var osc = context.createOscillator();
+
+
+  osc.connect(gain);
+  gain.connect(analyser);
+
+  osc.start();
+
+  var array = new Uint8Array(analyser.frequencyBinCount);
+
+  function getAnalyserData() {
+    gain.gain.setValueAtTime(currentGain, context.currentTime);
+    analyser.getByteTimeDomainData(array);
+    var inrange = true;
+    var max = -1;
+    for (var i = 0; i < array.length; i++) {
+      if (array[i] > max) {
+        max = Math.abs(array[i] - 128);
+      }
+    }
+    if (max <= currentGain * 128) {
+      ok(true, "Analyser got scaled data for " + currentGain);
+      currentGain = tests.shift();
+      if (currentGain == undefined) {
+        SimpleTest.finish();
+        return;
+      }
+    }
+    requestAnimationFrame(getAnalyserData);
+  }
+
+  var tests = [1.0, 0.5, 0.0];
+  var currentGain = tests.shift();
+  requestAnimationFrame(getAnalyserData);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNode.html b/dom/media/webaudio/test/test_audioBufferSourceNode.html
new file mode 100644
index 0000000000..875c96c366
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNode.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.start(0);
+    source.buffer = buffer;
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var buffers = [];
+    var buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      buffer.getChannelData(1)[i] = buffer.getChannelData(0)[i];
+    }
+    buffers.push(buffer);
+    buffers.push(getEmptyBuffer(context, 2048));
+    return buffers;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html b/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html
new file mode 100644
index 0000000000..4d06c26cad
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode when an AudioBuffer's getChanneData buffer is detached</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createGarbage() {
+  var s = [];
+  for (var i = 0; i < 10000000; ++i) {
+    s.push(i);
+  }
+  var sum = 0;
+  for (var i = 0; i < s.length; ++i) {
+    sum += s[i];
+  }
+  return sum;
+}
+
+var worker = new Worker("audioBufferSourceNodeDetached_worker.js");
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 10000000, context.sampleRate);
+    var data = buffer.getChannelData(0);
+    for (var i = 0; i < data.length; ++i) {
+      data[i] = (i%100)/100 - 0.5;
+    }
+
+    // Detach the buffer now
+    var data = buffer.getChannelData(0).buffer;
+    worker.postMessage(data, [data]);
+    // Create garbage and GC to replace the buffer data with garbage
+    SpecialPowers.gc();
+    createGarbage();
+    SpecialPowers.gc();
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.start();
+    // This should play silence
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html b/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html
new file mode 100644
index 0000000000..08616bea6c
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ended event on AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+
+  source.onended = function(e) {
+    is(e.target, source, "Correct target for the ended event");
+    SimpleTest.finish();
+  };
+
+  source.start(0);
+  source.buffer = buffer;
+  source.connect(context.destination);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html
new file mode 100644
index 0000000000..0893cf10b3
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    // silence for half of the buffer, ones after that.
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 1024; i < 2048; i++) {
+      buffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+
+    // we start at the 1024 frames, we should only have ones.
+    source.loop = true;
+    source.loopStart = 1024 / context.sampleRate;
+    source.loopEnd = 2048 / context.sampleRate;
+    source.buffer = buffer;
+    source.start(0, 1024 / context.sampleRate, 1024 / context.sampleRate);
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 0; i < 4096; i++) {
+      expectedBuffer.getChannelData(0)[i] = 1;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html
new file mode 100644
index 0000000000..79c78dfe0f
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048 * 4,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    source.start(0);
+    source.loop = true;
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 4, context.sampleRate);
+    for (var i = 0; i < 4; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(0)[i * 2048 + j] = Math.sin(440 * 2 * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html
new file mode 100644
index 0000000000..6f60762eb8
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048 * 4,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.start(0);
+    source.loop = true;
+    source.loopStart = buffer.duration * 0.25;
+    source.loopEnd = buffer.duration * 0.75;
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 4, context.sampleRate);
+    for (var i = 0; i < 1536; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 512; j < 1536; ++j) {
+        expectedBuffer.getChannelData(0)[1536 + i * 1024 + j - 512] = Math.sin(440 * 2 * Math.PI * j / context.sampleRate);
+      }
+    }
+    for (var j = 7680; j < 2048 * 4; ++j) {
+      expectedBuffer.getChannelData(0)[j] = Math.sin(440 * 2 * Math.PI * (j - 7168) / context.sampleRate);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html
new file mode 100644
index 0000000000..eca4bf636b
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    source.loop = true;
+    source.loopStart = source.loopEnd = 1 / context.sampleRate;
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    return buffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html b/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
new file mode 100644
index 0000000000..89340ade8b
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode when start() is not called</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    var data = buffer.getChannelData(0);
+    for (var i = 0; i < data.length; ++i) {
+      data[i] = (i%100)/100 - 0.5;
+    }
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html b/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html
new file mode 100644
index 0000000000..6ca771af1d
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    source.start(0);
+    source.buffer = null;
+    is(source.buffer, null, "Try playing back a null buffer");
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html b/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html
new file mode 100644
index 0000000000..b7a16634e2
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the offset property on AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var fuzz = 0.3;
+
+if (navigator.platform.startsWith("Mac")) {
+  // bug 895720
+  fuzz = 0.6;
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var samplesFromSource = 0;
+  var context = new AudioContext();
+  var sp = context.createScriptProcessor(256);
+
+  sp.onaudioprocess = function(e) {
+    samplesFromSource += e.inputBuffer.length;
+  }
+
+  var buffer = context.createBuffer(1, context.sampleRate, context.sampleRate);
+  for (var i = 0; i < context.sampleRate; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+
+  source.onended = function(e) {
+    // The timing at which the audioprocess and ended listeners are called can
+    // change, hence the fuzzy equal here.
+    var errorRatio = samplesFromSource / (0.5 * context.sampleRate);
+    ok(errorRatio > (1.0 - fuzz) && errorRatio < (1.0 + fuzz),
+       "Correct number of samples received (expected: " +
+       (0.5 * context.sampleRate) + ", actual: " + samplesFromSource + ").");
+    SimpleTest.finish();
+  };
+
+  source.buffer = buffer;
+  source.connect(sp);
+  source.start(0, 0.5);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html b/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html
new file mode 100644
index 0000000000..5088f1637c
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    source.buffer = buffer;
+
+    var srcWrapped = SpecialPowers.wrap(source);
+    ok("passThrough" in srcWrapped, "AudioBufferSourceNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [expectedBuffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html b/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html
new file mode 100644
index 0000000000..2cdcd72702
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var rate = 44100;
+var off = new OfflineAudioContext(1, rate, rate);
+var off2 = new OfflineAudioContext(1, rate, rate);
+
+var source = off.createBufferSource();
+var source2 = off2.createBufferSource();
+
+// a buffer of a 440Hz at half the length. If we detune by -1200 or set the
+// playbackRate to 0.5, we should get 44100 samples back with a sine at 220Hz.
+var buf = off.createBuffer(1, rate / 2, rate);
+var bufarray = buf.getChannelData(0);
+for (var i = 0; i < bufarray.length; i++) {
+  bufarray[i] = Math.sin(i * 440 * 2 * Math.PI / rate);
+}
+
+source.buffer = buf;
+source.playbackRate.value = 0.5; // 50% slowdown
+source.connect(off.destination);
+source.start(0);
+
+source2.buffer = buf;
+source2.detune.value = -1200; // one octave -> 50% slowdown
+source2.connect(off2.destination);
+source2.start(0);
+
+var finished = 0;
+function finish() {
+  finished++;
+  if (finished == 2) {
+    SimpleTest.finish();
+  }
+}
+
+off.startRendering().then((renderedPlaybackRate) => {
+  // we don't care about comparing the value here, we just want to know whether
+  // the second part is noisy.
+  var rmsValue = rms(renderedPlaybackRate, 0, 22050);
+  ok(rmsValue != 0, "Resampling happened (rms of the second part " + rmsValue + ")");
+
+  off2.startRendering().then((renderedDetune) => {
+    var rmsValue = rms(renderedDetune, 0, 22050);
+    ok(rmsValue != 0, "Resampling happened (rms of the second part " + rmsValue + ")");
+    // The two buffers should be the same: detune of -1200 is a 50% slowdown
+    compareBuffers(renderedPlaybackRate, renderedDetune);
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html b/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html
new file mode 100644
index 0000000000..269d5380e7
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html
@@ -0,0 +1,410 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test suspend, resume and close method of the AudioContext</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function tryToCreateNodeOnClosedContext(ctx) {
+  ok(ctx.state, "closed", "The context is in closed state");
+
+  [ { name: "createBufferSource" },
+    { name: "createMediaStreamDestination",
+      onOfflineAudioContext: false},
+    { name: "createScriptProcessor" },
+    { name: "createStereoPanner" },
+    { name: "createAnalyser" },
+    { name: "createGain" },
+    { name: "createDelay" },
+    { name: "createBiquadFilter" },
+    { name: "createWaveShaper" },
+    { name: "createPanner" },
+    { name: "createConvolver" },
+    { name: "createChannelSplitter" },
+    { name: "createChannelMerger" },
+    { name: "createDynamicsCompressor" },
+    { name: "createOscillator" },
+    { name: "createMediaElementSource",
+      args: [new Audio()],
+      onOfflineAudioContext: false },
+    { name: "createMediaStreamSource",
+      args: [new Audio().mozCaptureStream()],
+      onOfflineAudioContext: false } ].forEach(function(e) {
+
+      if (e.onOfflineAudioContext == false &&
+          ctx instanceof OfflineAudioContext) {
+        return;
+      }
+
+      expectException(function() {
+        ctx[e.name].apply(ctx, e.args);
+      }, DOMException.INVALID_STATE_ERR);
+    });
+}
+
+function loadFile(url, callback) {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", url, true);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    callback(xhr.response);
+  };
+  xhr.send();
+}
+
+// createBuffer, createPeriodicWave and decodeAudioData should work on a context
+// that has `state` == "closed"
+function tryLegalOpeerationsOnClosedContext(ctx) {
+  ok(ctx.state, "closed", "The context is in closed state");
+
+  [ { name: "createBuffer",
+      args: [1, 44100, 44100] },
+    { name: "createPeriodicWave",
+      args: [new Float32Array(10), new Float32Array(10)] }
+  ].forEach(function(e) {
+    expectNoException(function() {
+      ctx[e.name].apply(ctx, e.args);
+    });
+  });
+  loadFile("ting-44.1k-1ch.ogg", function(buf) {
+    ctx.decodeAudioData(buf).then(function(decodedBuf) {
+      ok(true, "decodeAudioData on a closed context should work, it did.")
+      todo(false, "0 " + (ctx instanceof OfflineAudioContext ? "Offline" : "Realtime"));
+      finish();
+    }).catch(function(e){
+      ok(false, "decodeAudioData on a closed context should work, it did not");
+      finish();
+    });
+  });
+}
+
+// Test that MediaStreams that are the output of a suspended AudioContext are
+// producing silence
+// ac1 produce a sine fed to a MediaStreamAudioDestinationNode
+// ac2 is connected to ac1 with a MediaStreamAudioSourceNode, and check that
+// there is silence when ac1 is suspended
+function testMultiContextOutput() {
+  var ac1 = new AudioContext(),
+      ac2 = new AudioContext();
+
+  ac1.onstatechange = function() {
+    ac1.onstatechange = null;
+
+    var osc1 = ac1.createOscillator(),
+        mediaStreamDestination1 = ac1.createMediaStreamDestination();
+
+    var mediaStreamAudioSourceNode2 =
+      ac2.createMediaStreamSource(mediaStreamDestination1.stream),
+      sp2 = ac2.createScriptProcessor(),
+      silentBuffersInARow = 0;
+
+
+    sp2.onaudioprocess = function(e) {
+      ac1.suspend().then(function() {
+        is(ac1.state, "suspended", "ac1 is suspended");
+        sp2.onaudioprocess = checkSilence;
+      });
+      sp2.onaudioprocess = null;
+    }
+
+    function checkSilence(e) {
+      var input = e.inputBuffer.getChannelData(0);
+      var silent = true;
+      for (var i = 0; i < input.length; i++) {
+        if (input[i] != 0.0) {
+          silent = false;
+        }
+      }
+
+      todo(false, "input buffer is " + (silent ? "silent" : "noisy"));
+
+      if (silent) {
+        silentBuffersInARow++;
+        if (silentBuffersInARow == 10) {
+          ok(true,
+              "MediaStreams produce silence when their input is blocked.");
+          sp2.onaudioprocess = null;
+          ac1.close();
+          ac2.close();
+          todo(false,"1");
+          finish();
+        }
+      } else {
+        is(silentBuffersInARow, 0,
+            "No non silent buffer inbetween silent buffers.");
+      }
+    }
+
+    osc1.connect(mediaStreamDestination1);
+
+    mediaStreamAudioSourceNode2.connect(sp2);
+    osc1.start();
+  }
+}
+
+
+// Test that there is no buffering between contexts when connecting a running
+// AudioContext to a suspended AudioContext. Our ScriptProcessorNode does some
+// buffering internally, so we ensure this by using a very very low frequency
+// on a sine, and oberve that the phase has changed by a big enough margin.
+function testMultiContextInput() {
+  var ac1 = new AudioContext(),
+      ac2 = new AudioContext();
+
+  ac1.onstatechange = function() {
+    ac1.onstatechange = null;
+
+    var osc1 = ac1.createOscillator(),
+        mediaStreamDestination1 = ac1.createMediaStreamDestination(),
+        sp1 = ac1.createScriptProcessor();
+
+    var mediaStreamAudioSourceNode2 =
+      ac2.createMediaStreamSource(mediaStreamDestination1.stream),
+      sp2 = ac2.createScriptProcessor(),
+      eventReceived = 0;
+
+
+    osc1.frequency.value = 0.0001;
+
+    function checkDiscontinuity(e) {
+      var inputBuffer = e.inputBuffer.getChannelData(0);
+      if (eventReceived++ == 3) {
+        var delta = Math.abs(inputBuffer[1] - sp2.value),
+            theoreticalIncrement = 2048 * 3 * Math.PI * 2 * osc1.frequency.value / ac1.sampleRate;
+        ok(delta >= theoreticalIncrement,
+            "Buffering did not occur when the context was suspended (delta:" + delta + " increment: " + theoreticalIncrement+")");
+        ac1.close();
+        ac2.close();
+        sp1.onaudioprocess = null;
+        sp2.onaudioprocess = null;
+        todo(false, "2");
+        finish();
+      }
+    }
+
+    sp2.onaudioprocess = function(e) {
+      var inputBuffer = e.inputBuffer.getChannelData(0);
+      sp2.value = inputBuffer[inputBuffer.length - 1];
+      ac2.suspend().then(function() {
+          ac2.resume().then(function() {
+            sp2.onaudioprocess = checkDiscontinuity;
+            });
+          });
+    }
+
+    osc1.connect(mediaStreamDestination1);
+    osc1.connect(sp1);
+
+    mediaStreamAudioSourceNode2.connect(sp2);
+    osc1.start();
+  }
+}
+
+// Test that ScriptProcessorNode's onaudioprocess don't get called while the
+// context is suspended/closed. It is possible that we get the handler called
+// exactly once after suspend, because the event has already been sent to the
+// event loop.
+function testScriptProcessNodeSuspended() {
+  var ac = new AudioContext();
+  var sp = ac.createScriptProcessor();
+  var remainingIterations = 30;
+  var afterResume = false;
+  ac.onstatechange = function() {
+    ac.onstatechange = null;
+    sp.onaudioprocess = function() {
+      ok(ac.state == "running", "If onaudioprocess is called, the context" +
+          " must be running (was " + ac.state + ", remainingIterations:" + remainingIterations +")");
+      remainingIterations--;
+      if (!afterResume) {
+        if (remainingIterations == 0) {
+          ac.suspend().then(function() {
+            ac.resume().then(function() {
+              remainingIterations = 30;
+              afterResume = true;
+            });
+          });
+        }
+      } else {
+        sp.onaudioprocess = null;
+        todo(false,"3");
+        finish();
+      }
+    }
+  }
+  sp.connect(ac.destination);
+}
+
+// Take an AudioContext, make sure it switches to running when the audio starts
+// flowing, and then, call suspend, resume and close on it, tracking its state.
+function testAudioContext() {
+  var ac = new AudioContext();
+  is(ac.state, "suspended", "AudioContext should start in suspended state.");
+  var stateTracker = {
+    previous: ac.state,
+     // no promise for the initial suspended -> running
+    initial: {  handler: false },
+    suspend: { promise: false, handler: false },
+    resume: { promise: false, handler: false },
+    close: { promise: false, handler: false }
+  };
+
+  function initialSuspendToRunning() {
+    ok(stateTracker.previous == "suspended" &&
+       ac.state == "running",
+       "AudioContext should switch to \"running\" when the audio hardware is" +
+       " ready.");
+
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterSuspend;
+    stateTracker.initial.handler = true;
+
+    ac.suspend().then(function() {
+      ok(!stateTracker.suspend.promise && !stateTracker.suspend.handler,
+        "Promise should be resolved before the callback, and only once.")
+      stateTracker.suspend.promise = true;
+    });
+  }
+
+  function afterSuspend() {
+    ok(stateTracker.previous == "running" &&
+       ac.state == "suspended",
+       "AudioContext should switch to \"suspend\" when the audio stream is" +
+       "suspended.");
+    ok(stateTracker.suspend.promise && !stateTracker.suspend.handler,
+        "Handler should be called after the callback, and only once");
+
+    stateTracker.suspend.handler = true;
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterResume;
+
+    ac.resume().then(function() {
+      ok(!stateTracker.resume.promise && !stateTracker.resume.handler,
+        "Promise should be called before the callback, and only once");
+      stateTracker.resume.promise = true;
+    });
+  }
+
+  function afterResume() {
+    ok(stateTracker.previous == "suspended" &&
+       ac.state == "running",
+   "AudioContext should switch to \"running\" when the audio stream resumes.");
+
+    ok(stateTracker.resume.promise && !stateTracker.resume.handler,
+       "Handler should be called after the callback, and only once");
+
+    stateTracker.resume.handler = true;
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterClose;
+
+    ac.close().then(function() {
+      ok(!stateTracker.close.promise && !stateTracker.close.handler,
+        "Promise should be called before the callback, and only once");
+      stateTracker.close.promise = true;
+      tryToCreateNodeOnClosedContext(ac);
+      tryLegalOpeerationsOnClosedContext(ac);
+    });
+  }
+
+  function afterClose() {
+    ok(stateTracker.previous == "running" &&
+       ac.state == "closed",
+       "AudioContext should switch to \"closed\" when the audio stream is" +
+       " closed.");
+    ok(stateTracker.close.promise && !stateTracker.close.handler,
+       "Handler should be called after the callback, and only once");
+  }
+
+  ac.onstatechange = initialSuspendToRunning;
+}
+
+function testOfflineAudioContext() {
+  var o = new OfflineAudioContext(1, 44100, 44100);
+  is(o.state, "suspended", "OfflineAudioContext should start in suspended state.");
+
+  expectRejectedPromise(o, "suspend", "NotSupportedError");
+  expectRejectedPromise(o, "resume", "NotSupportedError");
+  expectRejectedPromise(o, "close", "NotSupportedError");
+
+  var previousState = o.state,
+      finishedRendering = false;
+  function beforeStartRendering() {
+    ok(previousState == "suspended" && o.state == "running", "onstatechanged" +
+        "handler is called on state changed, and the new state is running");
+    previousState = o.state;
+    o.onstatechange = onRenderingFinished;
+  }
+
+  function onRenderingFinished() {
+    ok(previousState == "running" && o.state == "closed",
+        "onstatechanged handler is called when rendering finishes, " +
+        "and the new state is closed");
+    ok(finishedRendering, "The Promise that is resolved when the rendering is" +
+                    "done should be resolved earlier than the state change.");
+    previousState = o.state;
+    o.onstatechange = afterRenderingFinished;
+
+    tryToCreateNodeOnClosedContext(o);
+    tryLegalOpeerationsOnClosedContext(o);
+  }
+
+  function afterRenderingFinished() {
+    ok(false, "There should be no transition out of the closed state.");
+  }
+
+  o.onstatechange = beforeStartRendering;
+
+  o.startRendering().then(function(buffer) {
+    finishedRendering = true;
+  });
+}
+
+function testSuspendResumeEventLoop() {
+  var ac = new AudioContext();
+  var source = ac.createBufferSource();
+  source.buffer = ac.createBuffer(1, 44100, 44100);
+  source.onended = function() {
+    ok(true, "The AudioContext did resume.");
+    finish();
+  }
+  ac.onstatechange = function() {
+    ac.onstatechange = null;
+
+    ok(ac.state == "running", "initial state is running");
+    ac.suspend();
+    source.start();
+    ac.resume();
+  }
+}
+
+var remaining = 0;
+function finish() {
+  remaining--;
+  if (remaining == 0) {
+    SimpleTest.finish();
+  }
+}
+
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var tests = [
+    testAudioContext,
+    testOfflineAudioContext,
+    testScriptProcessNodeSuspended,
+    testMultiContextOutput,
+    testMultiContextInput,
+    testSuspendResumeEventLoop
+  ];
+  remaining = tests.length;
+  tests.forEach(function(f) { f() });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioDestinationNode.html b/dom/media/webaudio/test/test_audioDestinationNode.html
new file mode 100644
index 0000000000..b86c7169d0
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioDestinationNode.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioDestinationNode as EventTarget</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ac = new AudioContext()
+ac.destination.addEventListener("foo", function() {
+  ok(true, "Event received!");
+  SimpleTest.finish();
+}, false);
+ac.destination.dispatchEvent(new CustomEvent("foo"));
+
+</script>
+</pre>
+</body>
+</html>
+
diff --git a/dom/media/webaudio/test/test_audioParamChaining.html b/dom/media/webaudio/test/test_audioParamChaining.html
new file mode 100644
index 0000000000..6093e4425d
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamChaining.html
@@ -0,0 +1,77 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish()
+
+function frameToTime(frame, rate)
+{
+  return frame / rate;
+}
+
+const RATE = 44100;
+
+var oc = new OfflineAudioContext(1, 44100, RATE);
+// This allows us to have a source that is simply a DC offset.
+var source = oc.createBufferSource();
+var buf = oc.createBuffer(1, 1, RATE);
+buf.getChannelData(0)[0] = 1;
+source.loop = true;
+source.buffer = buf;
+
+source.start(0);
+
+var gain = oc.createGain();
+
+source.connect(gain).connect(oc.destination);
+
+var gain2 = oc.createGain();
+var rv2 = gain2.gain.linearRampToValueAtTime(0.1, 0.5);
+ok(rv2 instanceof AudioParam, "linearRampToValueAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "linearRampToValueAtTime returns the right AudioParam.");
+
+rv2 = gain2.gain.exponentialRampToValueAtTime(0.01, 1.0);
+ok(rv2 instanceof AudioParam,
+    "exponentialRampToValueAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain,
+    "exponentialRampToValueAtTime returns the right AudioParam.");
+
+rv2 = gain2.gain.setTargetAtTime(1.0, 2.0, 0.1);
+ok(rv2 instanceof AudioParam, "setTargetAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "setTargetAtTime returns the right AudioParam.");
+
+var array = new Float32Array(10);
+rv2 = gain2.gain.setValueCurveAtTime(array, 10, 11);
+ok(rv2 instanceof AudioParam, "setValueCurveAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "setValueCurveAtTime returns the right AudioParam.");
+
+// We chain three automation methods, making a gain step.
+var rv = gain.gain.setValueAtTime(0, frameToTime(0, RATE))
+                  .setValueAtTime(0.5, frameToTime(22000, RATE))
+                  .setValueAtTime(1, frameToTime(44000, RATE));
+
+ok(rv instanceof AudioParam, "setValueAtTime returns an AudioParam.");
+ok(rv == gain.gain, "setValueAtTime returns the right AudioParam.");
+
+oc.startRendering().then(function(rendered) {
+    console.log(rendered.getChannelData(0));
+    is(rendered.getChannelData(0)[0], 0,
+      "The value of the first step is correct.");
+    is(rendered.getChannelData(0)[22050], 0.5,
+      "The value of the second step is correct");
+    is(rendered.getChannelData(0)[44099], 1,
+      "The value of the third step is correct.");
+    SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamExponentialRamp.html b/dom/media/webaudio/test/test_audioParamExponentialRamp.html
new file mode 100644
index 0000000000..e1b1c5142a
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamExponentialRamp.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.exponentialRampToValue</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, 0);
+    gain.gain.exponentialRampToValueAtTime(V1, 2048/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V0 * Math.pow(V1 / V0, (t - T0) / (T1 - T0));
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamGain.html b/dom/media/webaudio/test/test_audioParamGain.html
new file mode 100644
index 0000000000..b971becce0
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamGain.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam with pre-gain </title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ctx = new AudioContext();
+var source = ctx.createOscillator();
+var lfo = ctx.createOscillator();
+var lfoIntensity = ctx.createGain();
+var effect = ctx.createGain();
+var sp = ctx.createScriptProcessor(2048, 1);
+
+source.frequency.value = 440;
+lfo.frequency.value = 2;
+// Very low gain, so the LFO should have very little influence
+// on the source, its RMS value should be close to the nominal value
+// for a sine wave.
+lfoIntensity.gain.value = 0.0001;
+
+lfo.connect(lfoIntensity);
+lfoIntensity.connect(effect.gain);
+source.connect(effect);
+effect.connect(sp);
+
+sp.onaudioprocess = function(e) {
+  var buffer = e.inputBuffer.getChannelData(0);
+  var rms = 0;
+  for (var i = 0; i < buffer.length; i++) {
+    rms += buffer[i] * buffer[i];
+  }
+
+  rms /= buffer.length;
+  rms = Math.sqrt(rms);
+
+  // 1 / Math.sqrt(2) is the theoretical RMS value for a sine wave.
+  ok(fuzzyCompare(rms, 1 / Math.sqrt(2)),
+      "Gain correctly applied to the AudioParam.");
+
+  ctx = null;
+  sp.onaudioprocess = null;
+  lfo.stop(0);
+  source.stop(0);
+
+  SimpleTest.finish();
+}
+
+lfo.start(0);
+source.start(0);
+
+</script>
+</pre>
+</body>
diff --git a/dom/media/webaudio/test/test_audioParamLinearRamp.html b/dom/media/webaudio/test/test_audioParamLinearRamp.html
new file mode 100644
index 0000000000..31f1d80d6c
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamLinearRamp.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, 0);
+    gain.gain.linearRampToValueAtTime(V1, 2048/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V0 + (V1 - V0) * ((t - T0) / (T1 - T0));
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html b/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html
new file mode 100644
index 0000000000..bcb655b52d
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createConstantSource();
+
+    var gain = context.createGain();
+    gain.gain.setValueCurveAtTime(this.curve, T0, this.duration);
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    this.duration = 1024 / context.sampleRate;
+    this.curve = new Float32Array([1.0, 0.5, 0.75, 0.25]);
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    var data = expectedBuffer.getChannelData(0);
+    var step = 1024 / 3;
+    for (var i = 0; i < 2048; ++i) {
+      if (i < step) {
+        data[i] = 1.0 - 0.5*i/step;
+      } else if (i < 2*step) {
+        data[i] = 0.5 + 0.25*(i - step)/step;
+      } else if (i < 3*step) {
+        data[i] = 0.75 - 0.5*(i - 2*step)/step;
+      } else {
+        data[i] = 0.25;
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetCurveAtTimeTwice.html b/dom/media/webaudio/test/test_audioParamSetCurveAtTimeTwice.html
new file mode 100644
index 0000000000..0f976380e0
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetCurveAtTimeTwice.html
@@ -0,0 +1,68 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setValueCurveAtTime twice</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+
+function linearInterpolate(t0, v0, t1, v1, t)
+{
+  return v0 + (v1 - v0) * ((t - t0) / (t1 - t0));
+}
+
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var curve2 = new Float32Array(100);
+    for (var i = 0; i < 100; ++i) {
+      curve2[i] = Math.sin(220 * 6 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createConstantSource();
+
+    var gain = context.createGain();
+    gain.gain.setValueCurveAtTime(curve2, T0, this.duration/2);
+    //Set a different curve from the first one
+    gain.gain.setValueCurveAtTime(this.curve, T0, this.duration);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    this.duration = 1024 / context.sampleRate;
+    this.curve = new Float32Array(100);
+    for (var i = 0; i < 100; ++i) {
+      this.curve[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      step = 1024.0/99.0;
+      var current = Math.floor(i / step);
+      var next = current + 1;
+      if (next < this.curve.length) {
+        expectedBuffer.getChannelData(0)[i] = linearInterpolate(current*step, this.curve[current], next*step, this.curve[next], i);
+      } else {
+        expectedBuffer.getChannelData(0)[i] = this.curve[99];
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetCurveAtTimeZeroDuration.html b/dom/media/webaudio/test/test_audioParamSetCurveAtTimeZeroDuration.html
new file mode 100644
index 0000000000..174c15c6fe
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetCurveAtTimeZeroDuration.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueCurveAtTime(this.curve, this.T0, 0);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    this.T0 = 1024 / context.sampleRate;
+    this.curve = new Float32Array(100);
+    for (var i = 0; i < 100; ++i) {
+      this.curve[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 1024; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 1;
+    }
+    for (var i = 1024; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = this.curve[99];
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html b/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html
new file mode 100644
index 0000000000..ccb35ca7b4
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 10;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = V0;
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V1 + (V0 - V1) * Math.exp(-(t - T0) / TimeConstant);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html b/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html
new file mode 100644
index 0000000000..bad12ca31b
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime with zero time constant</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = V0;
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V1;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetValueAtTime.html b/dom/media/webaudio/test/test_audioParamSetValueAtTime.html
new file mode 100644
index 0000000000..1ab5159351
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetValueAtTime.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = 0;
+    gain.gain.setValueAtTime(V0, 1024/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 1024; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 0.1;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html b/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html
new file mode 100644
index 0000000000..510beb3c70
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam timeline events scheduled after the destination stream has started playback</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.requestFlakyTimeout("This test needs to wait until the AudioDestinationNode's stream's timer starts.");
+
+var gTest = {
+  length: 16384,
+  numberOfChannels: 1,
+  createGraphAsync: function(context, callback) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    setTimeout(function() {
+      var source = context.createBufferSource();
+      source.buffer = sourceBuffer;
+      source.start(context.currentTime);
+      source.stop(context.currentTime + sourceBuffer.duration);
+
+      var gain = context.createGain();
+      gain.gain.setValueAtTime(0, context.currentTime);
+      gain.gain.setTargetAtTime(0, context.currentTime + sourceBuffer.duration, 1);
+      source.connect(gain);
+
+      callback(gain);
+    }, 100);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_badConnect.html b/dom/media/webaudio/test/test_badConnect.html
new file mode 100644
index 0000000000..b0d7c8f0cc
--- /dev/null
+++ b/dom/media/webaudio/test/test_badConnect.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context1 = new AudioContext();
+  var context2 = new AudioContext();
+
+  var destination1 = context1.destination;
+  var destination2 = context2.destination;
+
+  isnot(destination1, destination2, "Destination nodes should not be the same");
+  isnot(destination1.context, destination2.context, "Destination nodes should not have the same context");
+
+  var source1 = context1.createBufferSource();
+
+  expectException(function() {
+    source1.connect(destination1, 1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    source1.connect(destination1, 0, 1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    source1.connect(destination2);
+  }, DOMException.SYNTAX_ERR);
+
+  source1.connect(destination1);
+
+  expectException(function() {
+    source1.disconnect(1);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNode.html b/dom/media/webaudio/test/test_biquadFilterNode.html
new file mode 100644
index 0000000000..078f891798
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNode.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-3, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var filter = context.createBiquadFilter();
+
+  source.buffer = buffer;
+
+  source.connect(filter);
+  filter.connect(destination);
+
+  // Verify default values
+  is(filter.type, "lowpass", "Correct default value for type");
+  near(filter.frequency.defaultValue, 350, "Correct default value for filter frequency");
+  near(filter.detune.defaultValue, 0, "Correct default value for filter detune");
+  near(filter.Q.defaultValue, 1, "Correct default value for filter Q");
+  near(filter.gain.defaultValue, 0, "Correct default value for filter gain");
+  is(filter.channelCount, 2, "Biquad filter node has 2 input channels by default");
+  is(filter.channelCountMode, "max", "Correct channelCountMode for the biquad filter node");
+  is(filter.channelInterpretation, "speakers", "Correct channelCountInterpretation for the biquad filter node");
+
+  // Make sure that we can set all of the valid type values
+  var types = [
+    "lowpass",
+    "highpass",
+    "bandpass",
+    "lowshelf",
+    "highshelf",
+    "peaking",
+    "notch",
+    "allpass",
+  ];
+  for (var i = 0; i < types.length; ++i) {
+    filter.type = types[i];
+  }
+
+  // Make sure getFrequencyResponse handles invalid frequencies properly
+  var frequencies = new Float32Array([-1.0, context.sampleRate*0.5 - 1.0, context.sampleRate]);
+  var magResults = new Float32Array(3);
+  var phaseResults = new Float32Array(3);
+  filter.getFrequencyResponse(frequencies, magResults, phaseResults);
+  ok(isNaN(magResults[0]), "Invalid input frequency should give NaN magnitude response");
+  ok(!isNaN(magResults[1]), "Valid input frequency should not give NaN magnitude response");
+  ok(isNaN(magResults[2]), "Invalid input frequency should give NaN magnitude response");
+  ok(isNaN(phaseResults[0]), "Invalid input frquency should give NaN phase response");
+  ok(!isNaN(phaseResults[1]), "Valid input frquency should not give NaN phase response");
+  ok(isNaN(phaseResults[2]), "Invalid input frquency should give NaN phase response");
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+    filter.disconnect();
+
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html b/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html
new file mode 100644
index 0000000000..59fc8ab4f5
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var filter = context.createBiquadFilter();
+
+    source.buffer = this.buffer;
+
+    source.connect(filter);
+
+    var filterWrapped = SpecialPowers.wrap(filter);
+    ok("passThrough" in filterWrapped, "BiquadFilterNode should support the passThrough API");
+    filterWrapped.passThrough = true;
+
+    source.start(0);
+    return filter;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html b/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html
new file mode 100644
index 0000000000..390f2cdb01
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode after a GainNode and tail - Bugs 924286 and 924288</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+
+var gTest = {
+  length: signalLength,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    // Two oscillators scheduled sequentially
+    var signalDuration = signalLength / context.sampleRate;
+    var osc1 = context.createOscillator();
+    osc1.type = "square";
+    osc1.start(0);
+    osc1.stop(signalDuration / 2);
+    var osc2 = context.createOscillator();
+    osc2.start(signalDuration / 2);
+    osc2.stop(signalDuration);
+
+    // Comparing a biquad on each source with one on both sources checks that
+    // the biquad on the first source doesn't shut down early.
+    var biquad1 = context.createBiquadFilter();
+    osc1.connect(biquad1);
+    var biquad2 = context.createBiquadFilter();
+    osc2.connect(biquad2);
+
+    var gain = context.createGain();
+    gain.gain.value = -1;
+    osc1.connect(gain);
+    osc2.connect(gain);
+
+    var biquadWithGain = context.createBiquadFilter();
+    gain.connect(biquadWithGain);
+
+    // The output of biquadWithGain should be the inverse of the sum of the
+    // outputs of biquad1 and biquad2, so blend them together and expect
+    // silence.
+    var blend = context.createGain();
+    biquad1.connect(blend);
+    biquad2.connect(blend);
+    biquadWithGain.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1027864.html b/dom/media/webaudio/test/test_bug1027864.html
new file mode 100644
index 0000000000..0c115d1a05
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1027864.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test bug 1027864</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+function observer(subject, topic, data) {
+  var id = parseInt(data);
+  var index = ids.indexOf(id);
+  if (index != -1) {
+    ok(true, "Dropping id " + id + " at index " + index);
+    ids.splice(index, 1);
+    if (ids.length == 0) {
+      SimpleTest.executeSoon(function() {
+        SimpleTest.finish();
+      });
+    }
+  }
+}
+
+function id(node) {
+  return SpecialPowers.getPrivilegedProps(node, "id");
+}
+
+SpecialPowers.addObserver(observer, "webaudio-node-demise", false);
+
+SimpleTest.registerCleanupFunction(function() {
+  SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+});
+
+var ac = new AudioContext();
+var ids;
+
+(function() {
+  var delay = ac.createDelay();
+  delay.delayTime.value = 0.03;
+
+  var gain = ac.createGain();
+  gain.gain.value = 0.6;
+
+  delay.connect(gain);
+  gain.connect(delay);
+
+  gain.connect(ac.destination);
+
+  var source = ac.createOscillator();
+
+  source.connect(gain);
+  source.start(ac.currentTime);
+  source.stop(ac.currentTime + 0.1);
+
+  ids = [ id(delay), id(gain), id(source) ];
+})();
+
+setInterval(function() {
+  forceCC();
+}, 200);
+
+function forceCC() {
+  SpecialPowers.DOMWindowUtils.cycleCollect();
+  SpecialPowers.DOMWindowUtils.garbageCollect();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1056032.html b/dom/media/webaudio/test/test_bug1056032.html
new file mode 100644
index 0000000000..98fb159f76
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1056032.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we can decode an mp3 (bug 1056032)</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var filename = "small-shot.mp3";
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", filename);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    var context = new AudioContext();
+    context.decodeAudioData(xhr.response, function(b) {
+      ok(true, "We can decode an mp3 using decodeAudioData");
+      SimpleTest.finish();
+    }, function() {
+      ok(false, "We should be able to decode an mp3 using decodeAudioData but couldn't");
+      SimpleTest.finish();
+    });
+  };
+  xhr.send(null);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1113634.html b/dom/media/webaudio/test/test_bug1113634.html
new file mode 100644
index 0000000000..8995589f32
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1113634.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime where the target time is the same as the time of a previous event</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 0.1;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, T0);
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V1 + (V0 - V1) * Math.exp(-(t - T0) / TimeConstant);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1118372.html b/dom/media/webaudio/test/test_bug1118372.html
new file mode 100644
index 0000000000..ca3fc6b0df
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1118372.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var context = new OfflineAudioContext(1, 2048, 44100);
+
+  var osc=context.createOscillator();
+  var gain=context.createGain();
+  var shaper=context.createWaveShaper();
+  gain.gain.value=0.1;
+  shaper.curve=new Float32Array([-0.5,-0.5,1,1]);
+
+  osc.connect(gain);
+  gain.connect(shaper);
+  shaper.connect(context.destination);
+  osc.start(0);
+
+  context.startRendering().then(function(buffer) {
+    var samples = buffer.getChannelData(0);
+    // the signal should be scaled
+    var failures = 0;
+    for (var i = 0; i < 2048; ++i) {
+      if (samples[i] > 0.5) {
+        failures = failures + 1;
+      }
+    }
+    ok(failures == 0, "signal should have been rescaled by gain: found " + failures + " points too loud.");
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1255618.html b/dom/media/webaudio/test/test_bug1255618.html
new file mode 100644
index 0000000000..246a044380
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1255618.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test sync XHR does not crash unlinked AudioContext</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script>
+SimpleTest.waitForExplicitFinish();
+
+const filename = "test_bug1255618.html";
+
+function collect_and_fetch() {
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+
+    var xhr = new XMLHttpRequest();
+    xhr.open("GET", filename, false);
+    var ended = false;
+    xhr.onloadend = function() { ended = true; }
+    // Sync XHR will suspend timeouts, which involves any AudioContexts still
+    // registered with the window.
+    // See https://bugzilla.mozilla.org/show_bug.cgi?id=1255618#c0
+    xhr.send(null);
+
+    ok(ended, "No crash during fetch");
+    SimpleTest.finish();
+}
+
+var ac = new AudioContext();
+
+ac.onstatechange = function () {
+    ac.onstatechange = null;
+    is(ac.state, "running", "statechange to running");
+    ac = null;
+    SimpleTest.executeSoon(collect_and_fetch);
+}
+
+</script>
+</body>
diff --git a/dom/media/webaudio/test/test_bug1267579.html b/dom/media/webaudio/test/test_bug1267579.html
new file mode 100644
index 0000000000..62eda14dce
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1267579.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test that PeriodicWave handles fundamental fequency of zero</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// This is the smallest value that the test framework will accept
+const testLength = 256;
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  runTest();
+});
+
+var gTest = {
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var osc = context.createOscillator();
+    osc.setPeriodicWave(context.
+                        createPeriodicWave(new Float32Array([0.0, 1.0]),
+                                           new Float32Array(2)));
+    osc.frequency.value = 0.0;
+    osc.start();
+    return osc;
+  },
+  createExpectedBuffers: function(context) {
+    var buffer = context.createBuffer(1, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+      buffer.getChannelData(0)[i] = 1.0;
+    }
+    return buffer;
+  },
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug808374.html b/dom/media/webaudio/test/test_bug808374.html
new file mode 100644
index 0000000000..cc4e02f418
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug808374.html
@@ -0,0 +1,22 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 808374</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+try {
+  var ctx = new AudioContext();
+  ctx.createBuffer(0, 1, ctx.sampleRate);
+} catch (e) {
+  ok(true, "The test should not crash during CC");
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug827541.html b/dom/media/webaudio/test/test_bug827541.html
new file mode 100644
index 0000000000..9940c112e0
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug827541.html
@@ -0,0 +1,22 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Tell the cycle collector about the audio contexts owned by nsGlobalWindow</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+  var iframe = document.createElementNS("http://www.w3.org/1999/xhtml", "iframe");
+  document.body.appendChild(iframe);
+  var frameWin = iframe.contentWindow;
+  new frameWin.AudioContext();
+  document.body.removeChild(iframe);
+  new frameWin.AudioContext();
+
+  ok(true, "This test should not leak");
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug839753.html b/dom/media/webaudio/test/test_bug839753.html
new file mode 100644
index 0000000000..bbab10b250
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug839753.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 839753</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).destination.expando = null;
+ok(true, "The test should not trigger wrapper cache assertions");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug845960.html b/dom/media/webaudio/test/test_bug845960.html
new file mode 100644
index 0000000000..4e37f91bf6
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug845960.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 845960</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).decodeAudioData(new ArrayBuffer(0), function() {});
+ok(true, "Should not crash when the optional failure callback is not specified");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug856771.html b/dom/media/webaudio/test/test_bug856771.html
new file mode 100644
index 0000000000..8a6e622c26
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug856771.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test for bug 856771</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var source = context.createBufferSource();
+  source.connect(context.destination);
+  ok(true, "Nothing should leak");
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug866570.html b/dom/media/webaudio/test/test_bug866570.html
new file mode 100644
index 0000000000..0a1feca61b
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug866570.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 859600</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).foo = null;
+ok(true, "The test should not fatally assert");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug866737.html b/dom/media/webaudio/test/test_bug866737.html
new file mode 100644
index 0000000000..40fcf83fdf
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug866737.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test for bug 866737</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var context = new AudioContext();
+
+(function() {
+  var d = context.createDelay();
+  var panner = context.createPanner();
+  d.connect(panner);
+  var gain = context.createGain();
+  panner.connect(gain);
+  gain.connect(context.destination);
+  gain.disconnect(0);
+})();
+
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+
+var gain = context.createGain();
+gain.connect(context.destination);
+gain.disconnect(0);
+
+ok(true, "No crashes should happen!");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867089.html b/dom/media/webaudio/test/test_bug867089.html
new file mode 100644
index 0000000000..650676a449
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867089.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867089</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+
+  // Test invalid playbackRate values for AudioBufferSourceNode.
+  var source = ctx.createBufferSource();
+  var buffer = ctx.createBuffer(2, 2048, 8000);
+  source.buffer = buffer;
+  source.playbackRate.value = 0.0;
+  source.connect(ctx.destination);
+  source.start(0);
+
+  var source2 = ctx.createBufferSource();
+  source2.buffer = buffer;
+  source2.playbackRate.value = -1.0;
+  source2.connect(ctx.destination);
+  source2.start(0);
+
+  var source3 = ctx.createBufferSource();
+  source3.buffer = buffer;
+  source3.playbackRate.value = 3000000.0;
+  source3.connect(ctx.destination);
+  source3.start(0);
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867104.html b/dom/media/webaudio/test/test_bug867104.html
new file mode 100644
index 0000000000..82852ba51b
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867104.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867104</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+  var source = ctx.createBufferSource();
+  var b0 = ctx.createBuffer(32,798,22050);
+  var b1 = ctx.createBuffer(32,28,22050);
+  var sp = ctx.createScriptProcessor(0, 2, 0);
+  source.buffer = b0;
+  source.connect(sp);
+  source.start(0);
+  source.buffer = b1;
+  sp.onaudioprocess = function() {
+    ok(true, "We did not crash.");
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  };
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867174.html b/dom/media/webaudio/test/test_bug867174.html
new file mode 100644
index 0000000000..ce88be1b32
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867174.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867174</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+
+  var source = ctx.createBufferSource();
+  var buffer = ctx.createBuffer(2, 2048, 8000);
+  source.playbackRate.setTargetAtTime(0, 2, 3);
+  var sp = ctx.createScriptProcessor();
+  source.connect(sp);
+  sp.connect(ctx.destination);
+  source.start(0);
+
+  sp.onaudioprocess = function(e) {
+    // Now set the buffer
+    source.buffer = buffer;
+
+    ok(true, "We did not crash.");
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  };
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867203.html b/dom/media/webaudio/test/test_bug867203.html
new file mode 100644
index 0000000000..0ca31263d3
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867203.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867203</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+
+  var panner1 = ctx.createPanner();
+  panner1.setVelocity(1, 1, 1);
+  ctx.listener.setVelocity(1, 1, 1);
+  (function() {
+    ctx.createBufferSource().connect(panner1);
+  })();
+  SpecialPowers.forceGC();
+  SpecialPowers.forceCC();
+  ctx.createPanner();
+
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug875221.html b/dom/media/webaudio/test/test_bug875221.html
new file mode 100644
index 0000000000..16560ae752
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug875221.html
@@ -0,0 +1,239 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 875221</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test is generated by a fuzzer, so we leave these setTimeouts untouched.");
+
+try { o0 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o0); } catch(e) {  }
+try { o1 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o1.listener.dopplerFactor = 1; } catch(e) {  }
+try { o2 = o1.createScriptProcessor(); } catch(e) {  }
+try { o3 = o1.createChannelMerger(4); } catch(e) {  }
+try { o1.listener.dopplerFactor = 3; } catch(e) {  }
+try { o1.listener.setPosition(0, 134217728, 64) } catch(e) {  }
+try { o1.listener.dopplerFactor = 15; } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o4 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o4.listener.speedOfSound = 2048; } catch(e) {  }
+try { o4.listener.setPosition(32768, 1, 1) } catch(e) {  }
+try { o5 = o1.createChannelSplitter(4); } catch(e) {  }
+try { o4.listener.setVelocity(4, 1, 0) } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.listener.setPosition(64, 1, 0) } catch(e) {  }
+try { o1.listener.setOrientation(4194304, 15, 8388608, 15, 1, 1) } catch(e) {  }
+try { o1.listener.dopplerFactor = 256; } catch(e) {  }
+try { o6 = o4.createDelay(16); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.listener.setOrientation(0, 1, 0, 0, 31, 1073741824) } catch(e) {  }
+try { o4.listener.speedOfSound = 1; } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o6.connect(o3, 1, 0) } catch(e) {  }
+try { o1.listener.setPosition(4294967296, 32, 1) } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.listener.setVelocity(1, 256, 0) } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o3.disconnect() } catch(e) {  }
+setTimeout("try { o4.startRendering(); } catch(e) {  }",50)
+try { o4.listener.setOrientation(0, 0, 2048, 128, 16384, 127) } catch(e) {  }
+try { o4.listener.setVelocity(0, 4, 1) } catch(e) {  }
+try { o7 = o4.createScriptProcessor(1024, 4, 1); } catch(e) {  }
+try { o8 = o4.createDynamicsCompressor(); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+SpecialPowers.forceCC();
+SpecialPowers.forceGC();
+try { o4.listener.setOrientation(8192, 1, 1, 512, 0, 15) } catch(e) {  }
+setTimeout("try { o7.onaudioprocess = function() {}; } catch(e) {  }",50)
+try { o1.startRendering(); } catch(e) {  }
+try { o1.listener.speedOfSound = 1073741824; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o9 = o4.createScriptProcessor(1024, 1, 4); } catch(e) {  }
+try { o10 = o4.createAnalyser(); } catch(e) {  }
+try { o4.listener.speedOfSound = 0; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o4.listener.setVelocity(524288, 1, 65536) } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  } setTimeout(done, 0);",1000)
+try { o7.connect(o4); } catch(e) {  }
+try { o1.listener.setVelocity(1, 127, 31) } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+setTimeout("try { o5.disconnect() } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+setTimeout("try { o1.listener.dopplerFactor = 1; } catch(e) {  }",100)
+try { o5.disconnect() } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o10.disconnect() } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o11 = o1.createGain(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setOrientation(31, 0, 15, 0, 33554432, 1) } catch(e) {  }
+try { o4.listener.dopplerFactor = 1; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o9.connect(o4); } catch(e) {  }",50)
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o9.connect(o1); } catch(e) {  }",200)
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o12 = o4.createDynamicsCompressor(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o9.onaudioprocess = function() {}; } catch(e) {  }
+try { o1.listener.speedOfSound = 262144; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",50)
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o13 = o4.createGain(); } catch(e) {  }
+try { o4.listener.dopplerFactor = 31; } catch(e) {  }
+try { o11.gain.value = 268435456; } catch(e) {  }
+try { o1.listener.setOrientation(63, 3, 1, 63, 1, 2147483648) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o4.listener.setVelocity(1, 0, 1) } catch(e) {  }
+try { o11.gain.value = 65536; } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",200)
+try { o14 = o4.createDynamicsCompressor(); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",50)
+try { o7.connect(o1); } catch(e) {  }
+try { o15 = o1.createWaveShaper(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o16 = o1.createWaveShaper(); } catch(e) {  }
+try { o11.gain.value = 1; } catch(e) {  }
+try { o1.listener.speedOfSound = 16; } catch(e) {  }
+try { o4.listener.setVelocity(0, 127, 15) } catch(e) {  }
+try { o1.listener.setVelocity(0, 2048, 16777216) } catch(e) {  }
+try { o13.gain.value = 0; } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o17 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o0); } catch(e) {  }
+try { o4.listener.setVelocity(3, 1, 256) } catch(e) {  }
+try { o11.gain.cancelScheduledValues(1) } catch(e) {  }
+try { o1.listener.dopplerFactor = 524288; } catch(e) {  }
+try { o9.onaudioprocess = function() {}; } catch(e) {  }
+setTimeout("try { o7.connect(o13, 0, 0) } catch(e) {  }",50)
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o10.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o1.listener.speedOfSound = 1; } catch(e) {  }
+try { o15.disconnect() } catch(e) {  }
+try { o11.gain.exponentialRampToValueAtTime(0, 15) } catch(e) {  }
+try { o15.curve = new Float32Array(15); } catch(e) {  }
+try { o4.listener.setVelocity(1, 1, 1) } catch(e) {  }
+try { o14.connect(o6, 0, 0) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o7.connect(o1); } catch(e) {  }",100)
+try { o4.listener.setVelocity(1, 7, 1) } catch(e) {  }
+try { o18 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o18); } catch(e) {  }
+try { o19 = o4.createGain(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o4.listener.dopplerFactor = 0; } catch(e) {  }
+try { o1.listener.setPosition(256, 16, 1) } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",50)
+try { o7.connect(o1); } catch(e) {  }
+try { o4.listener.speedOfSound = 31; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o9.connect(o4); } catch(e) {  }",1000)
+try { o11.gain.value = 127; } catch(e) {  }
+try { o7.connect(o7, 0, 0) } catch(e) {  }
+try { o4.listener.speedOfSound = 63; } catch(e) {  }
+try { o11.gain.value = 33554432; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.speedOfSound = 16; } catch(e) {  }
+try { o4.listener.setVelocity(1048576, 0, 127) } catch(e) {  }
+try { o1.listener.dopplerFactor = 0; } catch(e) {  }
+try { o6.connect(o2, 0, 1) } catch(e) {  }
+try { o5.disconnect() } catch(e) {  }
+try { o3.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o16.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o9.disconnect() } catch(e) {  }
+try { o4.listener.speedOfSound = 1; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o11.gain.setValueCurveAtTime(new Float32Array(3), 2048, 3) } catch(e) {  }
+try { o13.gain.value = 8; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setOrientation(1, 2048, 1, 1, 0, 31) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o1.listener.speedOfSound = 256; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setVelocity(1, 67108864, 128) } catch(e) {  }
+setTimeout("try { o1.listener.setVelocity(0, 1, 1) } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+setTimeout("try { o20 = o1.createBiquadFilter(); } catch(e) {  }",200)
+try { o13.gain.value = 4096; } catch(e) {  }
+try { o1.listener.dopplerFactor = 0; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",200)
+try { o7.connect(o1); } catch(e) {  }
+try { o3.connect(o15, 1, 1) } catch(e) {  }
+try { o2.connect(o12, 0, 0) } catch(e) {  }
+try { o19.gain.exponentialRampToValueAtTime(1, 0) } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+
+function done() {
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+}
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug875402.html b/dom/media/webaudio/test/test_bug875402.html
new file mode 100644
index 0000000000..2dc347fc1a
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug875402.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 875402</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+SimpleTest.requestFlakyTimeout("This test is generated by a fuzzer, so we leave these setTimeouts untouched.");
+
+try { o1 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o2 = o1.createScriptProcessor(); } catch(e) {  }
+try { o4 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o5 = o1.createChannelSplitter(4); } catch(e) {  }
+try { o7 = o4.createScriptProcessor(1024, 4, 1); } catch(e) {  }
+SpecialPowers.forceCC();
+SpecialPowers.forceGC();
+try { o1.startRendering(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o9 = o4.createScriptProcessor(1024, 1, 4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  } done();",1000)
+try { o7.connect(o4); } catch(e) {  }
+setTimeout("try { o5.disconnect() } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",50)
+try { o13 = o4.createGain(); } catch(e) {  }
+setTimeout("try { o7.connect(o13, 0, 0) } catch(e) {  }",50)
+
+function done() {
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug894150.html b/dom/media/webaudio/test/test_bug894150.html
new file mode 100644
index 0000000000..08fd724136
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug894150.html
@@ -0,0 +1,21 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script>
+
+var ac = new AudioContext();
+ac.createPanner();
+var listener = ac.listener;
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+listener.setOrientation(0, 0, -1, 0, 0, 0);
+
+ok(true, "No crashes should happen!");
+
+</script>
+</body>
diff --git a/dom/media/webaudio/test/test_bug956489.html b/dom/media/webaudio/test/test_bug956489.html
new file mode 100644
index 0000000000..920889290a
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug956489.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test when and currentTime are in the same coordinate system</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while for the AudioContext's timer to start.");
+addLoadEvent(function() {
+  var freq = 330;
+
+  var context = new AudioContext();
+
+  var buffer = context.createBuffer(1, context.sampleRate / freq, context.sampleRate);
+  for (var i = 0; i < buffer.length; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(2 * Math.PI * i / buffer.length);
+  }
+
+  var source = context.createBufferSource();
+  source.loop = true;
+  source.buffer = buffer;
+
+  setTimeout(function () {
+      var finished = false;
+
+      source.start(context.currentTime);
+      var processor = context.createScriptProcessor(256, 1, 1);
+      processor.onaudioprocess = function (e) {
+          if (finished) return;
+          var c = e.inputBuffer.getChannelData(0);
+          var result = true;
+
+          for (var i = 0; i < buffer.length; ++i) {
+              if (Math.abs(c[i] - buffer.getChannelData(0)[i]) > 1e-9) {
+                  result = false;
+                  break;
+              }
+          }
+          finished = true;
+          ok(result, "when and currentTime are in same time coordinate system");
+          SimpleTest.finish();
+      }
+      processor.connect(context.destination);
+      source.connect(processor);
+  }, 500);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug964376.html b/dom/media/webaudio/test/test_bug964376.html
new file mode 100644
index 0000000000..1d9af1c1e0
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug964376.html
@@ -0,0 +1,64 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test repeating audio is not distorted</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function gcd(a, b) {
+  if (b === 0) {
+    return a;
+  }
+  return gcd(b, a % b);
+}
+
+var SAMPLE_PLACEMENT = 128;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+
+  createGraph: function(context) {
+    var freq = Math.round(context.sampleRate / SAMPLE_PLACEMENT);
+    var dur = context.sampleRate / gcd(freq, context.sampleRate);
+    var buffer = context.createBuffer(1, dur, context.sampleRate);
+
+    for (var i = 0; i < context.sampleRate; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(freq * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.loop = true;
+    source.playbackRate.setValueAtTime(0.5, SAMPLE_PLACEMENT / context.sampleRate);
+    source.start(0);
+
+    return source;
+  },
+
+  createExpectedBuffers: function(context) {
+    var freq = Math.round(context.sampleRate / SAMPLE_PLACEMENT);
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    var c = expectedBuffer.getChannelData(0);
+    for (var i = 0; i < c.length; ++i) {
+      if (i < SAMPLE_PLACEMENT) {
+        c[i] = Math.sin(freq * 2 * Math.PI * i / context.sampleRate);
+      } else {
+        c[i] = Math.sin(freq / 2 * 2 * Math.PI * (i + SAMPLE_PLACEMENT) / context.sampleRate);
+      }
+    }
+
+    return expectedBuffer;
+  },
+};
+
+runTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug966247.html b/dom/media/webaudio/test/test_bug966247.html
new file mode 100644
index 0000000000..9224ac2d46
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug966247.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether an audio file played with a volume set to 0 plays silence</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<audio preload=none src="ting-48k-1ch.ogg" controls> </audio>
+<script>
+  SimpleTest.waitForExplicitFinish();
+
+  var count = 20;
+
+  function isSilent(b) {
+    for (var i = 0; i < b.length; b++) {
+      if (b[i] != 0.0) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  var a = document.getElementsByTagName("audio")[0];
+  a.volume = 0.0;
+  var ac = new AudioContext();
+  var measn = ac.createMediaElementSource(a);
+  var sp = ac.createScriptProcessor();
+
+  sp.onaudioprocess = function(e) {
+    var inputBuffer = e.inputBuffer.getChannelData(0);
+    ok(isSilent(inputBuffer), "The volume is set to 0, so all the elements of the buffer are supposed to be equal to 0.0");
+  }
+  // Connect the MediaElementAudioSourceNode to the ScriptProcessorNode to check
+  // the audio volume.
+  measn.connect(sp);
+  a.play();
+
+  a.addEventListener("ended", function() {
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  });
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug972678.html b/dom/media/webaudio/test/test_bug972678.html
new file mode 100644
index 0000000000..d0cb4e4190
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug972678.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test buffers do not interfere when scheduled in sequence</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var OFFSETS = [0.005, 0.01, 0.02, 0.03];
+var LENGTH = 128;
+
+var gTest = {
+  length: 128 * OFFSETS.length,
+  numberOfChannels: 1,
+
+  createGraph: function(context) {
+    var gain = context.createGain();
+
+    // create a repeating sample
+    var repeatingSample = context.createBuffer(1, 2, context.sampleRate);
+    var c = repeatingSample.getChannelData(0);
+    for (var i = 0; i < repeatingSample.length; ++i) {
+      c[i] = i % 2 == 0 ? 1 : -1;
+    }
+
+    OFFSETS.forEach(function (offset, offsetIdx) {
+      // Schedule a set of nodes to repeat the sample.
+      for (var i = 0; i < LENGTH; i += repeatingSample.length) {
+        var source = context.createBufferSource();
+        source.buffer = repeatingSample;
+        source.connect(gain);
+        source.start((offsetIdx * LENGTH + i + offset) / context.sampleRate);
+      }
+
+      buffer = context.createBuffer(1, LENGTH, context.sampleRate);
+      c = buffer.getChannelData(0);
+      for (var i = 0; i < buffer.length; ++i) {
+        c[i] = i % 2 == 0 ? -1 : 1;
+      }
+
+      var source = context.createBufferSource();
+      source.buffer = buffer;
+      source.connect(gain);
+      source.start((offsetIdx * LENGTH + offset) / context.sampleRate);
+    });
+
+    return gain;
+  },
+
+  createExpectedBuffers: function(context) {
+    return context.createBuffer(1, gTest.length, context.sampleRate);
+  },
+};
+
+runTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelMergerNode.html b/dom/media/webaudio/test/test_channelMergerNode.html
new file mode 100644
index 0000000000..a76aaa2e87
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelMergerNode.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelMergerNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 6,
+  createGraph: function(context) {
+    var buffers = [];
+    for (var j = 0; j < 6; ++j) {
+      var buffer = context.createBuffer(2, 2048, context.sampleRate);
+      for (var i = 0; i < 2048; ++i) {
+        buffer.getChannelData(0)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+        // Second channel is silent
+      }
+      buffers.push(buffer);
+    }
+
+    var merger = context.createChannelMerger();
+    is(merger.channelCount, 1, "merger node has 1 input channels");
+    is(merger.channelCountMode, "explicit", "Correct channelCountMode for the merger node");
+    is(merger.channelInterpretation, "speakers", "Correct channelCountInterpretation for the merger node");
+
+    for (var i = 0; i < 6; ++i) {
+      var source = context.createBufferSource();
+      source.buffer = buffers[i];
+      source.connect(merger, 0, i);
+      source.start(0);
+    }
+
+    return merger;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(6, 2048, context.sampleRate);
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(i)[j] = 0.5 * Math.sin(440 * 2 * (i + 1) * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html b/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html
new file mode 100644
index 0000000000..22f0a39cb8
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelMergerNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 6,
+  createGraph: function(context) {
+    var buffers = [];
+    for (var j = 0; j < 6; ++j) {
+      var buffer = context.createBuffer(2, 2048, context.sampleRate);
+      for (var i = 0; i < 2048; ++i) {
+        buffer.getChannelData(0)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+        // Second channel is silent
+      }
+      buffers.push(buffer);
+    }
+
+    var merger = context.createChannelMerger();
+    is(merger.channelCount, 1, "merger node has 1 input channels");
+    is(merger.channelCountMode, "explicit", "Correct channelCountMode for the merger node");
+    is(merger.channelInterpretation, "speakers", "Correct channelCountInterpretation for the merger node");
+
+    for (var i = 0; i < 6; ++i) {
+      var source = context.createBufferSource();
+      source.buffer = buffers[i];
+      var gain = context.createGain();
+      gain.gain.value = 0.5;
+      source.connect(gain);
+      gain.connect(merger, 0, i);
+      source.start(0);
+    }
+
+    return merger;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(6, 2048, context.sampleRate);
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(i)[j] = 0.5 * 0.5 * Math.sin(440 * 2 * (i + 1) * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelSplitterNode.html b/dom/media/webaudio/test/test_channelSplitterNode.html
new file mode 100644
index 0000000000..30cb0028c8
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelSplitterNode.html
@@ -0,0 +1,71 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelSplitterNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the splitter node is special in that it creates multiple
+// output ports.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(4, 2048, context.sampleRate);
+  for (var j = 0; j < 4; ++j) {
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+    }
+  }
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var splitter = context.createChannelSplitter();
+  is(splitter.channelCount, 2, "splitter node has 2 input channels by default");
+  is(splitter.channelCountMode, "max", "Correct channelCountMode for the splitter node");
+  is(splitter.channelInterpretation, "speakers", "Correct channelCountInterpretation for the splitter node");
+
+  source.buffer = buffer;
+  source.connect(splitter);
+
+  var channelsSeen = 0;
+  function createHandler(i) {
+    return function(e) {
+      is(e.inputBuffer.numberOfChannels, 1, "Correct input channel count");
+      if (i < 4) {
+        compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(i));
+      } else {
+        compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+      }
+      e.target.onaudioprocess = null;
+      ++channelsSeen;
+
+      if (channelsSeen == 6) {
+        SimpleTest.finish();
+      }
+    };
+  }
+
+  for (var i = 0; i < 6; ++i) {
+    var sp = context.createScriptProcessor(2048, 1);
+    splitter.connect(sp, i);
+    sp.onaudioprocess = createHandler(i);
+    sp.connect(destination);
+  }
+
+  source.start(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html b/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html
new file mode 100644
index 0000000000..8e16271f36
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelSplitterNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the splitter node is special in that it creates multiple
+// output ports.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(4, 2048, context.sampleRate);
+  var expectedBuffer = context.createBuffer(4, 2048, context.sampleRate);
+  for (var j = 0; j < 4; ++j) {
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+      expectedBuffer.getChannelData(j)[i] = buffer.getChannelData(j)[i] / 2;
+    }
+  }
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var splitter = context.createChannelSplitter();
+  is(splitter.channelCount, 2, "splitter node has 2 input channels by default");
+  is(splitter.channelCountMode, "max", "Correct channelCountMode for the splitter node");
+  is(splitter.channelInterpretation, "speakers", "Correct channelCountInterpretation for the splitter node");
+
+  source.buffer = buffer;
+  var gain = context.createGain();
+  gain.gain.value = 0.5;
+  source.connect(gain);
+  gain.connect(splitter);
+
+  var channelsSeen = 0;
+  function createHandler(i) {
+    return function(e) {
+      is(e.inputBuffer.numberOfChannels, 1, "Correct input channel count");
+      if (i < 4) {
+        compareBuffers(e.inputBuffer.getChannelData(0), expectedBuffer.getChannelData(i));
+      } else {
+        compareBuffers(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+      }
+      e.target.onaudioprocess = null;
+      ++channelsSeen;
+
+      if (channelsSeen == 6) {
+        SimpleTest.finish();
+      }
+    };
+  }
+
+  for (var i = 0; i < 6; ++i) {
+    var sp = context.createScriptProcessor(2048, 1);
+    splitter.connect(sp, i);
+    sp.onaudioprocess = createHandler(i);
+    sp.connect(destination);
+  }
+
+  source.start(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNode.html b/dom/media/webaudio/test/test_convolverNode.html
new file mode 100644
index 0000000000..38b58bd9b7
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNode.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the ConvolverNode interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var conv = context.createConvolver();
+
+  is(conv.channelCount, 2, "Convolver node has 2 input channels by default");
+  is(conv.channelCountMode, "clamped-max", "Correct channelCountMode for the Convolver node");
+  is(conv.channelInterpretation, "speakers", "Correct channelCountInterpretation for the Convolver node");
+
+  is(conv.buffer, null, "Default buffer value");
+  conv.buffer = context.createBuffer(2, 1024, 22050);
+  is(conv.normalize, true, "Default normalize value");
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeChannelCount.html b/dom/media/webaudio/test/test_convolverNodeChannelCount.html
new file mode 100644
index 0000000000..d5b261c818
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeChannelCount.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode channel count</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const responseLength = 1000;
+const outputLength = 2048; // < signalLength + responseLength to test bug 910171
+
+var gTest = {
+  length: outputLength,
+  numberOfChannels: 2,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(2, signalLength, context.sampleRate);
+    for (var i = 0; i < signalLength; ++i) {
+      var sample = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      // When mixed into a single channel, this produces silence
+      buffer.getChannelData(0)[i] = sample;
+      buffer.getChannelData(1)[i] = -sample;
+    }
+
+    var response = context.createBuffer(2, responseLength, context.sampleRate);
+    for (var i = 0; i < responseLength; ++i) {
+      response.getChannelData(0)[i] = i / responseLength;
+      response.getChannelData(1)[i] = 1 - (i / responseLength);
+    }
+
+    var convolver = context.createConvolver();
+    convolver.buffer = response;
+    convolver.channelCount = 1;
+
+    expectException(function() { convolver.channelCount = 3; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    convolver.channelCountMode = "explicit";
+    expectException(function() { convolver.channelCountMode = "max"; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    convolver.channelInterpretation = "discrete";
+    convolver.channelInterpretation = "speakers";
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(convolver);
+    source.start(0);
+
+    return convolver;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeDelay.html b/dom/media/webaudio/test/test_convolverNodeDelay.html
new file mode 100644
index 0000000000..2e8caf8027
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeDelay.html
@@ -0,0 +1,72 @@
+<!DOCTYPE html>
+<title>Test convolution to delay a triangle pulse</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+const sampleRate = 48000;
+const LENGTH = 12800;
+// tolerate 16-bit math.
+const EPSILON = 1.0 / Math.pow(2, 15);
+
+// Triangle pulse
+var sourceBuffer = new OfflineAudioContext(1, 1, sampleRate).
+  createBuffer(1, 2 * 128, sampleRate);
+var channelData = sourceBuffer.getChannelData(0);
+for (var i = 0; i < 128; ++i) {
+  channelData[i] = i/128;
+  channelData[128 + i] = 1.0 - i/128;
+}
+
+function test_delay_index(delayIndex) {
+
+  var context = new OfflineAudioContext(2, LENGTH, sampleRate);
+
+  var merger = context.createChannelMerger(2);
+  merger.connect(context.destination);
+
+  var impulse = context.createBuffer(1, delayIndex + 1, sampleRate);
+  impulse.getChannelData(0)[delayIndex] = 1.0;
+  var convolver = context.createConvolver();
+  convolver.normalize = false;
+  convolver.buffer = impulse;
+  convolver.connect(merger, 0, 0);
+
+  var delayTime = delayIndex/sampleRate;
+  var delay = context.createDelay(delayTime || 1/sampleRate);
+  delay.delayTime.value = delayTime;
+  delay.connect(merger, 0, 1);
+
+  var source = context.createBufferSource();
+  source.buffer = sourceBuffer;
+  source.connect(convolver);
+  source.connect(delay);
+  source.start(0);
+
+  return context.startRendering().
+    then((buffer) => {
+      var convolverOutput = buffer.getChannelData(0);
+      var delayOutput = buffer.getChannelData(1);
+      var maxDiff = 0.0;
+      var maxIndex = 0;
+      for (var i = 0; i < buffer.length; ++i) {
+        var diff = Math.abs(convolverOutput[i] - delayOutput[i]);
+        if (diff > maxDiff) {
+          maxDiff = diff;
+          maxIndex = i;
+        }
+      }
+      // The convolver should produce similar output to the delay.
+      assert_approx_equals(convolverOutput[maxIndex], delayOutput[maxIndex],
+                           EPSILON, "output at " + maxIndex);
+    });
+}
+
+// The 5/4 ratio provides sampling across a range of delays and offsets within
+// blocks.
+for (var delayIndex = 0;
+     delayIndex < LENGTH;
+     delayIndex = Math.floor((5 * (delayIndex + 1)) / 4)) {
+  promise_test(test_delay_index.bind(null, delayIndex),
+               "Delay " + delayIndex);
+}
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html b/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html
new file mode 100644
index 0000000000..1cfb51ce8a
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html
@@ -0,0 +1,44 @@
+<!DOCTYPE html>
+<title>Test convolution effect has finite duration</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+promise_test(function() {
+
+  const responseLength = 256;
+  // Accept an influence period of twice the responseLength to accept FFT
+  // implementations.
+  const tolerancePeriod = 2 * responseLength;
+  const totalSize = tolerancePeriod + responseLength;
+
+  var context = new OfflineAudioContext(1, totalSize, 48000);
+
+  var responseBuffer =
+    context.createBuffer(1, responseLength, context.sampleRate);
+  var responseChannelData = responseBuffer.getChannelData(0);
+  responseChannelData[0] = 1;
+  responseChannelData[responseLength - 1] = 1;
+  var convolver = context.createConvolver();
+  convolver.buffer = responseBuffer;
+  convolver.connect(context.destination);
+
+  var sourceBuffer = context.createBuffer(1, totalSize, context.sampleRate);
+  sourceBuffer.getChannelData(0)[0] = NaN;
+  var source = context.createBufferSource();
+  source.buffer = sourceBuffer;
+  source.connect(convolver);
+  source.start();
+
+  return context.startRendering().
+    then((buffer) => {
+      var convolverOutput = buffer.getChannelData(0);
+      // There should be no non-zeros after the tolerance period.
+      var testIndex = tolerancePeriod;
+      for (;
+           testIndex < buffer.length - 1 && convolverOutput[testIndex] == 0;
+           ++testIndex) {
+      }
+      assert_equals(convolverOutput[testIndex], 0, "output at " + testIndex);
+    });
+});
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodePassThrough.html b/dom/media/webaudio/test/test_convolverNodePassThrough.html
new file mode 100644
index 0000000000..d5f9ef8ab5
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodePassThrough.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var convolver = context.createConvolver();
+
+    source.buffer = this.buffer;
+    convolver.buffer = this.buffer;
+
+    source.connect(convolver);
+
+    var convolverWrapped = SpecialPowers.wrap(convolver);
+    ok("passThrough" in convolverWrapped, "ConvolverNode should support the passThrough API");
+    convolverWrapped.passThrough = true;
+
+    source.start(0);
+    return convolver;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeWithGain.html b/dom/media/webaudio/test/test_convolverNodeWithGain.html
new file mode 100644
index 0000000000..7bbe240890
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeWithGain.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode after a GainNode - Bug 891254 </title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const responseLength = 100;
+const outputLength = 4096; // > signalLength + responseLength
+
+var gTest = {
+  length: outputLength,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, signalLength, context.sampleRate);
+    for (var i = 0; i < signalLength; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(2 * Math.PI * i / signalLength);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.start(0);
+
+    var response = context.createBuffer(1, responseLength, context.sampleRate);
+    for (var i = 0; i < responseLength; ++i) {
+      response.getChannelData(0)[i] = i / responseLength;
+    }
+
+    var gain = context.createGain();
+    gain.gain.value = -1;
+    source.connect(gain);
+
+    var convolver1 = context.createConvolver();
+    convolver1.buffer = response;
+    gain.connect(convolver1);
+
+    var convolver2 = context.createConvolver();
+    convolver2.buffer = response;
+    source.connect(convolver2);
+
+    // The output of convolver1 should be the inverse of convolver2, so blend
+    // them together and expect silence.
+    var blend = context.createGain();
+    convolver1.connect(blend);
+    convolver2.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNode_mono_mono.html b/dom/media/webaudio/test/test_convolverNode_mono_mono.html
new file mode 100644
index 0000000000..f7da2b0206
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNode_mono_mono.html
@@ -0,0 +1,73 @@
+<!DOCTYPE html>
+
+<html>
+<head>
+<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+<script type="text/javascript" src="webaudio.js"></script>
+<script type="text/javascript" src="layouttest-glue.js"></script>
+<script type="text/javascript" src="blink/audio-testing.js"></script>
+<script type="text/javascript" src="blink/convolution-testing.js"></script>
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+
+<body>
+
+<div id="description"></div>
+<div id="console"></div>
+
+<script>
+description("Tests ConvolverNode processing a mono channel with mono impulse response.");
+SimpleTest.waitForExplicitFinish();
+
+// To test the convolver, we convolve two square pulses together to
+// produce a triangular pulse.  To verify the result is correct we
+// check several parts of the result.  First, we make sure the initial
+// part of the result is zero (due to the latency in the convolver).
+// Next, the triangular pulse should match the theoretical result to
+// within some roundoff.  After the triangular pulse, the result
+// should be exactly zero, but round-off prevents that.  We make sure
+// the part after the pulse is sufficiently close to zero.  Finally,
+// the result should be exactly zero because the inputs are exactly
+// zero.
+function runTest() {
+    if (window.testRunner) {
+        testRunner.dumpAsText();
+        testRunner.waitUntilDone();
+    }
+
+    window.jsTestIsAsync = true;
+
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var squarePulse = createSquarePulseBuffer(context, pulseLengthFrames);
+    var trianglePulse = createTrianglePulseBuffer(context, 2 * pulseLengthFrames);
+
+    var bufferSource = context.createBufferSource();
+    bufferSource.buffer = squarePulse;
+
+    var convolver = context.createConvolver();
+    convolver.normalize = false;
+    convolver.buffer = squarePulse;
+
+    bufferSource.connect(convolver);
+    convolver.connect(context.destination);
+
+    bufferSource.start(0);
+
+    context.oncomplete = checkConvolvedResult(trianglePulse);
+    context.startRendering();
+}
+
+function finishJSTest() {
+  SimpleTest.finish();
+}
+
+runTest();
+successfullyParsed = true;
+
+</script>
+
+<script src="../fast/js/resources/js-test-post.js"></script>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_currentTime.html b/dom/media/webaudio/test/test_currentTime.html
new file mode 100644
index 0000000000..bb015e5e25
--- /dev/null
+++ b/dom/media/webaudio/test/test_currentTime.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioContext.currentTime</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while for the AudioContext's timer to start.");
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  is(ac.currentTime, 0, "AudioContext.currentTime should be 0 initially");
+  setTimeout(function() {
+    ok(ac.currentTime > 0, "AudioContext.currentTime should have increased by now");
+    SimpleTest.finish();
+  }, 1000);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeAudioDataPromise.html b/dom/media/webaudio/test/test_decodeAudioDataPromise.html
new file mode 100644
index 0000000000..d07f559360
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeAudioDataPromise.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the decodeAudioData API with Promise</title>
+
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  <script src="webaudio.js"></script>
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+var finished = 0;
+
+function finish() {
+  if (++finished == 2) {
+    SimpleTest.finish();
+  }
+}
+
+var ac = new AudioContext();
+// Test that a the promise is rejected with an invalid source buffer.
+expectNoException(function() {
+  var p = ac.decodeAudioData(" ");
+  ok(p instanceof Promise, "AudioContext.decodeAudioData should return a Promise");
+  p.then(function(data) {
+    ok(false, "Promise should not resolve with an invalid source buffer.");
+    finish();
+  }).catch(function(e) {
+    ok(true, "Promise should be rejected with an invalid source buffer.");
+    ok(e.name == "TypeError", "The error should be TypeError");
+    finish();
+  })
+});
+
+// Test that a the promise is resolved with a valid source buffer.
+var xhr = new XMLHttpRequest();
+xhr.open("GET", "ting-44.1k-1ch.ogg", true);
+xhr.responseType = "arraybuffer";
+xhr.onload = function() {
+  var p = ac.decodeAudioData(xhr.response);
+  ok(p instanceof Promise, "AudioContext.decodeAudioData should return a Promise");
+  p.then(function(data) {
+    ok(data instanceof AudioBuffer, "Promise should resolve, passing an AudioBuffer");
+    ok(true, "Promise should resolve with a valid source buffer.");
+    finish();
+  }).catch(function() {
+    ok(false, "Promise should not be rejected with a valid source buffer.");
+    finish();
+  });
+};
+xhr.send();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeMultichannel.html b/dom/media/webaudio/test/test_decodeMultichannel.html
new file mode 100644
index 0000000000..0fb2f5b3c6
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeMultichannel.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we can decode 4 channel wave file in webaudio, but not in &lt;audio&gt;</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var filename = "audio-quad.wav";
+
+SimpleTest.waitForExplicitFinish();
+
+function finishTest(a) {
+  if (a) {
+    a = null;
+    SimpleTest.finish();
+  }
+}
+
+function decodeUsingAudioElement() {
+  var a = new Audio();
+  a.addEventListener("error", function() {
+    ok(false, "Error loading metadata");
+    finishTest(a);
+  });
+  a.addEventListener("loadedmetadata", function() {
+    ok(true, "Metadata Loaded");
+    finishTest(a);
+  });
+
+  a.src = filename;
+  a.load();
+}
+
+addLoadEvent(function() {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", filename);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    var context = new AudioContext();
+    context.decodeAudioData(xhr.response, function(b) {
+      ok(true, "Decoding of a wave file with four channels succeded.");
+      is(b.numberOfChannels, 4, "The AudioBuffer should have 4 channels.");
+      decodeUsingAudioElement();
+    }, function() {
+      ok(false, "Decoding of a wave file with four channels failed.");
+      decodeUsingAudioElement();
+    });
+  };
+  xhr.send(null);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeOpusTail.html b/dom/media/webaudio/test/test_decodeOpusTail.html
new file mode 100644
index 0000000000..b5b53f685d
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeOpusTail.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Regression test to check that opus files don't have a tail at the end.</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+// This gets a 1 second Opus file and decodes it to a buffer. The opus file is
+// decoded at 48kHz, and the OfflineAudioContext is also at 48kHz, no resampling
+// is taking place.
+fetch('sweep-300-330-1sec.opus')
+.then(function(response) { return response.arrayBuffer(); })
+.then(function(buffer) {
+  var off = new OfflineAudioContext(1, 128, 48000);
+  off.decodeAudioData(buffer, function(decoded) {
+    var pcm = decoded.getChannelData(0);
+    is(pcm.length, 48000, "The length of the decoded file is correct.");
+    SimpleTest.finish();
+  });
+});
+
+</script>
diff --git a/dom/media/webaudio/test/test_delayNode.html b/dom/media/webaudio/test/test_delayNode.html
new file mode 100644
index 0000000000..a3e314ef79
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNode.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay();
+
+    source.buffer = buffer;
+
+    source.connect(delay);
+
+    ok(delay.delayTime, "The audioparam member must exist");
+    is(delay.delayTime.value, 0, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+    delay.delayTime.value = 0.5;
+    is(delay.delayTime.value, 0.5, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+    is(delay.channelCount, 2, "delay node has 2 input channels by default");
+    is(delay.channelCountMode, "max", "Correct channelCountMode for the delay node");
+    is(delay.channelInterpretation, "speakers", "Correct channelCountInterpretation for the delay node");
+
+    expectException(function() {
+      context.createDelay(0);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      context.createDelay(180);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectTypeError(function() {
+      context.createDelay(NaN);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      context.createDelay(-1);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    context.createDelay(1); // should not throw
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 2, context.sampleRate);
+    for (var i = 2048; i < 2048 * 2; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i - 2048) / context.sampleRate);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeAtMax.html b/dom/media/webaudio/test/test_delayNodeAtMax.html
new file mode 100644
index 0000000000..6c7dde3d18
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeAtMax.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode with maxDelayTime delay - bug 890528</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const delayLength = 1000; // Not on a block boundary
+const outputLength = 4096 // > signalLength + 2 * delayLength;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+var gTest = {
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, signalLength, context.sampleRate);
+    applySignal(buffer, 0);
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    const delayTime = delayLength / context.sampleRate;
+    var delay = context.createDelay(delayTime);
+    delay.delayTime.value = delayTime;
+
+    source.connect(delay);
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, outputLength, context.sampleRate);
+    applySignal(expectedBuffer, delayLength);
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeChannelChanges.html b/dom/media/webaudio/test/test_delayNodeChannelChanges.html
new file mode 100644
index 0000000000..229bfd0699
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeChannelChanges.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>test DelayNode channel count changes</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestCompleteLog();
+
+const bufferSize = 4096;
+
+var ctx;
+var testDelay;
+var stereoDelay;
+var invertor;
+
+function compareOutputs(callback) {
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  testDelay.connect(processor);
+  invertor.connect(processor);
+  processor.onaudioprocess =
+    function(e) {
+      compareBuffers(e.inputBuffer,
+                     ctx.createBuffer(2, bufferSize, ctx.sampleRate));
+      e.target.onaudioprocess = null;
+      callback();
+    }
+}
+
+function startTest() {
+  // And a two-channel signal
+  var merger = ctx.createChannelMerger();
+  merger.connect(testDelay);
+  merger.connect(stereoDelay);
+  var oscL = ctx.createOscillator();
+  oscL.connect(merger, 0, 0);
+  oscL.start(0);
+  var oscR = ctx.createOscillator();
+  oscR.type = "sawtooth";
+  oscR.connect(merger, 0, 1);
+  oscR.start(0);
+
+  compareOutputs(
+    function () {
+      // Disconnect the two-channel signal and test again
+      merger.disconnect();
+      compareOutputs(SimpleTest.finish);
+    });
+}
+
+function prepareTest() {
+  ctx = new AudioContext();
+
+  // The output of a test delay node with mono and stereo input will be
+  // compared with that of separate mono and stereo delay nodes.
+  const delayTime = 0.3 * bufferSize / ctx.sampleRate;
+  testDelay = ctx.createDelay(delayTime);
+  testDelay.delayTime.value = delayTime;
+  monoDelay = ctx.createDelay(delayTime);
+  monoDelay.delayTime.value = delayTime;
+  stereoDelay = ctx.createDelay(delayTime);
+  stereoDelay.delayTime.value = delayTime;
+
+  // Create a one-channel signal and connect to the delay nodes
+  var monoOsc = ctx.createOscillator();
+  monoOsc.frequency.value = 110;
+  monoOsc.connect(testDelay);
+  monoOsc.connect(monoDelay);
+  monoOsc.start(0);
+
+  // Invert the expected so that mixing with the test will find the difference.
+  invertor = ctx.createGain();
+  invertor.gain.value = -1.0;
+  monoDelay.connect(invertor);
+  stereoDelay.connect(invertor);
+
+  // Start the test after the delay nodes have begun processing.
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.connect(ctx.destination);
+
+  processor.onaudioprocess =
+    function(e) {
+      e.target.onaudioprocess = null;
+      processor.disconnect();
+      startTest();
+    };
+}
+prepareTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeCycles.html b/dom/media/webaudio/test/test_delayNodeCycles.html
new file mode 100644
index 0000000000..f5f2e67864
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeCycles.html
@@ -0,0 +1,157 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the support of cycles.</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const sampleRate = 48000;
+const inputLength = 2048;
+
+addLoadEvent(function() {
+  function addSine(b) {
+    for (var i = 0; i < b.length; i++) {
+      b[i] += Math.sin(440 * 2 * Math.PI * i / sampleRate);
+    }
+  }
+
+  function getSineBuffer(ctx) {
+    var buffer = ctx.createBuffer(1, inputLength, ctx.sampleRate);
+    addSine(buffer.getChannelData(0));
+    return buffer;
+  }
+
+  function createAndPlayWithCycleAndDelayNode(ctx, delayFrames) {
+    var source = ctx.createBufferSource();
+    source.buffer = getSineBuffer(ctx);
+
+    var gain = ctx.createGain();
+    var delay = ctx.createDelay();
+    delay.delayTime.value = delayFrames/ctx.sampleRate;
+
+    source.connect(gain);
+    gain.connect(delay);
+    delay.connect(ctx.destination);
+    // cycle
+    delay.connect(gain);
+
+    source.start(0);
+  }
+
+  function createAndPlayWithCycleAndNoDelayNode(ctx) {
+    var source = ctx.createBufferSource();
+    source.loop = true;
+    source.buffer = getSineBuffer(ctx);
+
+    var gain = ctx.createGain();
+    var gain2 = ctx.createGain();
+
+    source.connect(gain);
+    gain.connect(gain2);
+    // cycle
+    gain2.connect(gain);
+    gain2.connect(ctx.destination);
+
+    source.start(0);
+  }
+
+  function createAndPlayWithCycleAndNoDelayNodeInCycle(ctx) {
+    var source = ctx.createBufferSource();
+    source.loop = true;
+    source.buffer = getSineBuffer(ctx);
+
+    var delay = ctx.createDelay();
+    var gain = ctx.createGain();
+    var gain2 = ctx.createGain();
+
+    // Their is a cycle, a delay, but the delay is not in the cycle.
+    source.connect(delay);
+    delay.connect(gain);
+    gain.connect(gain2);
+    // cycle
+    gain2.connect(gain);
+    gain2.connect(ctx.destination);
+
+    source.start(0);
+  }
+
+  var remainingTests = 0;
+  function finish() {
+    if (--remainingTests == 0) {
+      SimpleTest.finish();
+    }
+  }
+
+  function getOfflineContext(oncomplete) {
+    var ctx = new OfflineAudioContext(1, sampleRate, sampleRate);
+    ctx.oncomplete = oncomplete;
+    return ctx;
+  }
+
+  function checkSilentBuffer(e) {
+    var buffer = e.renderedBuffer.getChannelData(0);
+    for (var i = 0; i < buffer.length; i++) {
+      if (buffer[i] != 0.0) {
+        ok(false, "buffer should be silent.");
+        finish();
+        return;
+      }
+    }
+    ok(true, "buffer should be silent.");
+    finish();
+  }
+
+  function checkNoisyBuffer(e, aDelayFrames) {
+    delayFrames = Math.max(128, aDelayFrames);
+
+    var expected = new Float32Array(e.renderedBuffer.length);
+    for (var i = delayFrames; i < expected.length; i += delayFrames) {
+      addSine(expected.subarray(i, i + inputLength));
+    }
+
+    compareChannels(e.renderedBuffer.getChannelData(0), expected);
+    finish();
+  }
+
+  function expectSilentOutput(f) {
+    remainingTests++;
+    var ctx = getOfflineContext(checkSilentBuffer);
+    f(ctx);
+    ctx.startRendering();
+  }
+
+  function expectNoisyOutput(delayFrames) {
+    remainingTests++;
+    var ctx = getOfflineContext();
+    ctx.oncomplete = function(e) { checkNoisyBuffer(e, delayFrames); };
+    createAndPlayWithCycleAndDelayNode(ctx, delayFrames);
+    ctx.startRendering();
+  }
+
+  // This is trying to make a graph with a cycle and no DelayNode in the graph.
+  // The cycle subgraph should be muted, in this graph the output should be silent.
+  expectSilentOutput(createAndPlayWithCycleAndNoDelayNode);
+  // This is trying to make a graph with a cycle and a DelayNode in the graph, but
+  // not part of the cycle.
+  // The cycle subgraph should be muted, in this graph the output should be silent.
+  expectSilentOutput(createAndPlayWithCycleAndNoDelayNodeInCycle);
+  // Those are making legal graphs, with at least one DelayNode in the cycle.
+  // There should be some non-silent output.
+  expectNoisyOutput(sampleRate/4);
+  // DelayNode.delayTime will be clamped to 128/ctx.sampleRate.
+  // There should be some non-silent output.
+  expectNoisyOutput(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodePassThrough.html b/dom/media/webaudio/test/test_delayNodePassThrough.html
new file mode 100644
index 0000000000..4945ee95c7
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodePassThrough.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay();
+
+    source.buffer = this.buffer;
+
+    source.connect(delay);
+
+    delay.delayTime.value = 0.5;
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    var delayWrapped = SpecialPowers.wrap(delay);
+    ok("passThrough" in delayWrapped, "DelayNode should support the passThrough API");
+    delayWrapped.passThrough = true;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    var silence = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [this.buffer, silence];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html b/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html
new file mode 100644
index 0000000000..b9cee458d7
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay(0.02);
+
+    source.buffer = this.buffer;
+
+    source.connect(delay);
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailIncrease.html b/dom/media/webaudio/test/test_delayNodeTailIncrease.html
new file mode 100644
index 0000000000..7516028245
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailIncrease.html
@@ -0,0 +1,71 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test increasing delay of DelayNode after input finishes</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 100;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+const delayBufferCount = 50;
+const delayLength = delayBufferCount * bufferSize + 700;
+
+var count = 0;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function onAudioProcess(e) {
+  switch(count) {
+  case 5:
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    break;
+  case delayBufferCount:
+    var offset = delayLength - count * bufferSize;
+    var ctx = e.target.context;
+    var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+    applySignal(expected, offset);
+    compareBuffers(e.inputBuffer, expected);
+    SimpleTest.finish();
+  }
+  count++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.onaudioprocess = onAudioProcess;
+
+  // Switch on delay at a time in the future.
+  var delayDuration = delayLength / ctx.sampleRate;
+  var delayStartTime = (delayLength - bufferSize) / ctx.sampleRate;
+  var delay = ctx.createDelay(delayDuration);
+  delay.delayTime.setValueAtTime(delayDuration, delayStartTime);
+  delay.connect(processor);
+
+  // Short signal that finishes before switching to long delay
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  var source = ctx.createBufferSource();
+  source.buffer = buffer;
+  source.start();
+  source.connect(delay);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html b/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html
new file mode 100644
index 0000000000..fa431d61b2
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode after input is disconnected</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// Web Audio doesn't provide a means to precisely time disconnect()s but we
+// can test that the output of delay nodes matches the output from their
+// sources before they are disconnected.
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 128;
+const bufferSize = 4096;
+const sourceCount = bufferSize / signalLength;
+// Delay should be long enough to allow CC to run
+var delayBufferCount = 20;
+const delayLength = delayBufferCount * bufferSize;
+
+var sourceOutput = new Float32Array(bufferSize);
+var delayOutputCount = 0;
+var sources = [];
+
+function onDelayOutput(e) {
+  if (delayOutputCount < delayBufferCount) {
+    delayOutputCount++;
+    return;
+  }
+
+  compareChannels(e.inputBuffer.getChannelData(0), sourceOutput);
+  e.target.onaudioprocess = null;
+  SimpleTest.finish();
+}
+
+function onSourceOutput(e) {
+  // Record the first buffer
+  e.inputBuffer.copyFromChannel(sourceOutput, 0);
+  e.target.onaudioprocess = null;
+}
+
+function disconnectSources() {
+  for (var i = 0; i < sourceCount; ++i) {
+    sources[i].disconnect();
+  }
+
+  SpecialPowers.forceGC();
+  SpecialPowers.forceCC();
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+
+  var sourceProcessor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  sourceProcessor.onaudioprocess = onSourceOutput;
+  // Keep audioprocess events going after source disconnect.
+  sourceProcessor.connect(ctx.destination);
+
+  var delayProcessor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  delayProcessor.onaudioprocess = onDelayOutput;
+
+  var delayDuration = delayLength / ctx.sampleRate;
+  for (var i = 0; i < sourceCount; ++i) {
+    var delay = ctx.createDelay(delayDuration);
+    delay.delayTime.value = delayDuration;
+    delay.connect(delayProcessor);
+
+    var source = ctx.createOscillator();
+    source.frequency.value = 440 + 10 * i
+    source.start(i * signalLength / ctx.sampleRate);
+    source.stop((i + 1) * signalLength / ctx.sampleRate);
+    source.connect(delay);
+    source.connect(sourceProcessor);
+
+    sources[i] = source;
+  }
+
+  // Assuming the above Web Audio operations have already scheduled an event
+  // to run in stable state and start the graph thread, schedule a subsequent
+  // event to disconnect the sources, which will remove main thread connection
+  // references before it knows the graph thread has started using the source
+  // streams.
+  SimpleTest.executeSoon(disconnectSources);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithGain.html b/dom/media/webaudio/test/test_delayNodeTailWithGain.html
new file mode 100644
index 0000000000..6994a7f9d9
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithGain.html
@@ -0,0 +1,72 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode indirectly connected to source</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 130;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+const delayBufferCount = 50;
+const delayLength = delayBufferCount * bufferSize + 700;
+
+var count = 0;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function onAudioProcess(e) {
+  switch(count) {
+  case 5:
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    break;
+  case delayBufferCount:
+    var offset = delayLength - count * bufferSize;
+    var ctx = e.target.context;
+    var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+    applySignal(expected, offset);
+    compareBuffers(e.inputBuffer, expected);
+    SimpleTest.finish();
+  }
+  count++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.onaudioprocess = onAudioProcess;
+
+  var delayDuration = delayLength / ctx.sampleRate;
+  var delay = ctx.createDelay(delayDuration);
+  delay.delayTime.value = delayDuration;
+  delay.connect(processor);
+
+  var gain = ctx.createGain();
+  gain.connect(delay);
+
+  // Short signal that finishes before garbage collection
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  var source = ctx.createBufferSource();
+  source.buffer = buffer;
+  source.start();
+  source.connect(gain);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html b/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html
new file mode 100644
index 0000000000..6c1cda5805
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html
@@ -0,0 +1,136 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode after input finishes and new input added</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// The buffer source will start on a block boundary, so keeping the signal
+// within one block ensures that it will not cross AudioProcessingEvent buffer
+// boundaries.
+const signalLength = 128;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+var delayBufferCount = 50;
+var delayBufferOffset;
+const delayLength = delayBufferCount * bufferSize;
+
+var phase = "initial";
+var sourceCount = 0;
+var delayCount = 0;
+var oscillator;
+var delay;
+var source;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function bufferIsSilent(buffer, out) {
+  for (var i = 0; i < buffer.length; ++i) {
+    if (buffer.getChannelData(0)[i] != 0) {
+      if (out) {
+        out.soundOffset = i;
+      }
+      return false;
+    }
+  }
+  return true;
+}
+
+function onDelayOutput(e) {
+  switch(phase) {
+
+  case "initial":
+    // Wait for oscillator sound to exit delay
+    if (bufferIsSilent(e.inputBuffer))
+      break;
+
+    phase = "played oscillator";
+    break;
+
+  case "played oscillator":
+    // First tail time has expired.  Start second source and remove references
+    // to the delay and connected second source.
+    oscillator.disconnect();
+    source.connect(delay);
+    source.start();
+    source = null;
+    delay = null;
+    phase = "started second source";
+    break;
+
+  case "second tail time":
+    if (delayCount == delayBufferCount) {
+      var ctx = e.target.context;
+      var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+      applySignal(expected, delayBufferOffset);
+      compareBuffers(e.inputBuffer, expected);
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+  }
+
+  delayCount++;
+}
+
+function onSourceOutput(e) {
+  switch(phase) {
+  case "started second source":
+    var out = {};
+    if (!bufferIsSilent(e.inputBuffer, out)) {
+      delayBufferCount += sourceCount;
+      delayBufferOffset = out.soundOffset;
+      phase = "played second source";
+    }
+    break;
+  case "played second source":
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    phase = "second tail time";
+    e.target.onaudioprocess = null;
+  }
+
+  sourceCount++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var delayDuration = delayLength / ctx.sampleRate;
+  delay = ctx.createDelay(delayDuration);
+  delay.delayTime.value = delayDuration;
+  var processor1 = ctx.createScriptProcessor(bufferSize, 1, 0);
+  delay.connect(processor1);
+  processor1.onaudioprocess = onDelayOutput;
+
+  // Signal to trigger initial tail time reference
+  oscillator = ctx.createOscillator();
+  oscillator.start(0);
+  oscillator.stop(100/ctx.sampleRate);
+  oscillator.connect(delay);
+
+  // Short signal, not started yet, with a ScriptProcessor to detect when it
+  // starts.  It should finish before garbage collection.
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  source = ctx.createBufferSource();
+  source.buffer = buffer;
+  var processor2 = ctx.createScriptProcessor(bufferSize, 1, 0);
+  source.connect(processor2);
+  processor2.onaudioprocess = onSourceOutput;
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeWithGain.html b/dom/media/webaudio/test/test_delayNodeWithGain.html
new file mode 100644
index 0000000000..768bea77cf
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeWithGain.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode with a GainNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay();
+
+    source.buffer = buffer;
+
+    var gain = context.createGain();
+    gain.gain.value = 0.5;
+
+    source.connect(gain);
+    gain.connect(delay);
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 2, context.sampleRate);
+    for (var i = 2048; i < 2048 * 2; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i - 2048) / context.sampleRate) / 2;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectAll.html b/dom/media/webaudio/test/test_disconnectAll.html
new file mode 100644
index 0000000000..9d3af066ea
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAll.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+          var data = sourceBuffer.getChannelData(0);
+          for (var j = 0; j < data.length; j++) {
+              data[j] = 1;
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var gain1 = context.createGain();
+          var gain2 = context.createGain();
+          var gain3 = context.createGain();
+          var merger = context.createChannelMerger(3);
+
+          source.connect(gain1);
+          source.connect(gain2);
+          source.connect(gain3);
+          gain1.connect(merger);
+          gain2.connect(merger);
+          gain3.connect(merger);
+          source.start();
+
+          source.disconnect();
+
+          return merger;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_disconnectAudioParam.html b/dom/media/webaudio/test/test_disconnectAudioParam.html
new file mode 100644
index 0000000000..1f4e79c564
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAudioParam.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioParam</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+          var data = sourceBuffer.getChannelData(0);
+          for (var j = 0; j < data.length; j++) {
+              data[j] = 1;
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var half = context.createGain();
+          var gain1 = context.createGain();
+          var gain2 = context.createGain();
+
+          half.gain.value = 0.5;
+
+          source.connect(gain1);
+          gain1.connect(gain2);
+          source.connect(half);
+
+          half.connect(gain1.gain);
+          half.connect(gain2.gain);
+
+          half.disconnect(gain2.gain);
+
+          source.start();
+
+          return gain2;
+        },
+        createExpectedBuffers: function(context) {
+          expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+          for (var i = 0; i < 256; ++i) {
+            expectedBuffer.getChannelData(0)[i] = 1.5;
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html b/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html
new file mode 100644
index 0000000000..a08ffa53bc
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioParam</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(2, 256, context.sampleRate);
+          for (var i = 1; i <= 2; i++) {
+            var data = sourceBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+                data[j] = i;
+            }
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var half = context.createGain();
+          var gain1 = context.createGain();
+          var gain2 = context.createGain();
+          var splitter = context.createChannelSplitter(2);
+
+          half.gain.value = 0.5;
+
+          source.connect(gain1);
+          gain1.connect(gain2);
+          source.connect(half);
+          half.connect(splitter);
+          splitter.connect(gain1.gain, 0);
+          splitter.connect(gain2.gain, 1);
+
+          splitter.disconnect(gain2.gain, 1);
+
+          source.start();
+
+          return gain2;
+        },
+        createExpectedBuffers: function(context) {
+          var expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+          for (var i = 1; i <= 2; i++) {
+            var data = expectedBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+              data[j] = (i == 1) ? 1.5 : 3.0;
+            }
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectExceptions.html b/dom/media/webaudio/test/test_disconnectExceptions.html
new file mode 100644
index 0000000000..ceba972c99
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectExceptions.html
@@ -0,0 +1,75 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var ctx = new AudioContext();
+      var sourceBuffer = ctx.createBuffer(2, 256, ctx.sampleRate);
+      for (var i = 1; i <= 2; i++) {
+        var data = sourceBuffer.getChannelData(i-1);
+        for (var j = 0; j < data.length; j++) {
+            data[j] = i;
+        }
+      }
+
+      var source = ctx.createBufferSource();
+      source.buffer = sourceBuffer;
+
+      var gain1 = ctx.createGain();
+      var splitter = ctx.createChannelSplitter(2);
+      var merger = ctx.createChannelMerger(2);
+      var gain2 = ctx.createGain();
+      var gain3 = ctx.createGain();
+
+      gain1.connect(splitter);
+      splitter.connect(gain2, 0);
+      splitter.connect(gain3, 1);
+      splitter.connect(merger, 0, 0);
+      splitter.connect(merger, 1, 1);
+      gain2.connect(gain3);
+      gain3.connect(ctx.destination);
+      merger.connect(ctx.destination);
+
+      expectException(function() {
+          splitter.disconnect(2);
+      }, DOMException.INDEX_SIZE_ERR);
+
+      expectNoException(function() {
+        splitter.disconnect(1);
+        splitter.disconnect(1);
+      });
+
+      expectException(function() {
+        gain1.disconnect(gain2);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        gain1.disconnect(gain3);
+        ok(false, 'Should get InvalidAccessError exception');
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain2, 2);
+      }, DOMException.INDEX_SIZE_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain1, 0);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain3, 0, 0);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(merger, 3, 0);
+      }, DOMException.INDEX_SIZE_ERR);
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNode.html b/dom/media/webaudio/test/test_disconnectFromAudioNode.html
new file mode 100644
index 0000000000..9311951466
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNode.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+          var data = sourceBuffer.getChannelData(0);
+          for (var j = 0; j < data.length; j++) {
+              data[j] = 1;
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var gain1 = context.createGain();
+          var gain2 = context.createGain();
+          var gain3 = context.createGain();
+
+          source.connect(gain1);
+          source.connect(gain2);
+
+          gain1.connect(gain3);
+          gain2.connect(gain3);
+
+          source.start();
+
+          source.disconnect(gain2);
+
+          return gain3;
+        },
+        createExpectedBuffers: function(context) {
+          expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+          for (var i = 0; i < 256; ++i) {
+            expectedBuffer.getChannelData(0)[i] = 1.0;
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html
new file mode 100644
index 0000000000..5c4e3ee5d0
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(2, 256, context.sampleRate);
+          for (var i = 1; i <= 2; i++) {
+            var data = sourceBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+                data[j] = i;
+            }
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var splitter = context.createChannelSplitter(2);
+          var gain1 = context.createGain();
+          var gain2 = context.createGain();
+          var merger = context.createChannelMerger(2);
+
+          source.connect(splitter);
+          splitter.connect(gain1, 0);
+          splitter.connect(gain2, 0);
+          splitter.connect(gain2, 1);
+          gain1.connect(merger, 0, 1);
+          gain2.connect(merger, 0, 1);
+          source.start();
+
+          splitter.disconnect(gain2, 0);
+
+          return merger;
+        },
+        createExpectedBuffers: function(context) {
+          expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+          for (var i = 0; i < 256; ++i) {
+            expectedBuffer.getChannelData(0)[i] = 0;
+            expectedBuffer.getChannelData(1)[i] = 3;
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html
new file mode 100644
index 0000000000..6526cf65b0
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 3,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(3, 256, context.sampleRate);
+          for (var i = 1; i <= 3; i++) {
+            var data = sourceBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+                data[j] = i;
+            }
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var splitter = context.createChannelSplitter(3);
+          var merger = context.createChannelMerger(3);
+
+          source.connect(splitter);
+          splitter.connect(merger, 0, 0);
+          splitter.connect(merger, 1, 1);
+          splitter.connect(merger, 2, 2);
+          source.start();
+
+          splitter.disconnect(merger, 2, 2);
+
+          return merger;
+        },
+        createExpectedBuffers: function(context) {
+          var expectedBuffer = context.createBuffer(3, 256, context.sampleRate);
+          for (var i = 1; i <= 3; i++) {
+            var data = expectedBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+              data[j] = (i == 3) ? 0 : i;
+            }
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html
new file mode 100644
index 0000000000..746b7ba938
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>
+      Test whether we can disconnect all outbound connection of an AudioNode
+    </title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+          var data = sourceBuffer.getChannelData(0);
+          for (var j = 0; j < data.length; j++) {
+              data[j] = 1;
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var merger = context.createChannelMerger(2);
+          var gain = context.createGain();
+
+          source.connect(merger, 0, 0);
+          source.connect(gain);
+          source.connect(merger, 0, 1);
+
+          source.disconnect(merger);
+
+          source.start();
+
+          return merger;
+        },
+        createExpectedBuffers: function(context) {
+          expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+          for (var channel = 0; channel < 2; channel++) {
+            for (var i = 0; i < 256; ++i) {
+              expectedBuffer.getChannelData(0)[i] = 0;
+            }
+          }
+
+          return expectedBuffer;
+        }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromOutput.html b/dom/media/webaudio/test/test_disconnectFromOutput.html
new file mode 100644
index 0000000000..8a6daf5c7d
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromOutput.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph: function(context) {
+          var sourceBuffer = context.createBuffer(3, 256, context.sampleRate);
+          for (var i = 1; i <= 3; i++) {
+            var data = sourceBuffer.getChannelData(i-1);
+            for (var j = 0; j < data.length; j++) {
+                data[j] = i;
+            }
+          }
+
+          var source = context.createBufferSource();
+          source.buffer = sourceBuffer;
+
+          var splitter = context.createChannelSplitter(3);
+          var sum = context.createGain();
+
+          source.connect(splitter);
+          splitter.connect(sum, 0);
+          splitter.connect(sum, 1);
+          splitter.connect(sum, 2);
+          source.start();
+
+          splitter.disconnect(1);
+
+          return sum;
+        },
+        createExpectedBuffers: function(context) {
+          expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+          for (var i = 0; i < 256; ++i) {
+            expectedBuffer.getChannelData(0)[i] = 4;
+          }
+
+          return expectedBuffer;
+        },
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNode.html b/dom/media/webaudio/test/test_dynamicsCompressorNode.html
new file mode 100644
index 0000000000..052b276714
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNode.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DynamicsCompressorNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-4, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var osc = context.createOscillator();
+  var sp = context.createScriptProcessor();
+
+  var compressor = context.createDynamicsCompressor();
+
+  osc.connect(compressor);
+  osc.connect(sp);
+  compressor.connect(context.destination);
+
+  is(compressor.channelCount, 2, "compressor node has 2 input channels by default");
+  is(compressor.channelCountMode, "explicit", "Correct channelCountMode for the compressor node");
+  is(compressor.channelInterpretation, "speakers", "Correct channelCountInterpretation for the compressor node");
+
+  // Verify default values
+  with (compressor) {
+    ok(threshold instanceof AudioParam, "treshold is an AudioParam");
+    near(threshold.defaultValue, -24, "Correct default value for threshold");
+    ok(knee instanceof AudioParam, "knee is an AudioParam");
+    near(knee.defaultValue, 30, "Correct default value for knee");
+    ok(ratio instanceof AudioParam, "knee is an AudioParam");
+    near(ratio.defaultValue, 12, "Correct default value for ratio");
+    is(typeof reduction, "number", "reduction is a number");
+    near(reduction, 0, "Correct default value for reduction");
+    ok(attack instanceof AudioParam, "attack is an AudioParam");
+    near(attack.defaultValue, 0.003, "Correct default value for attack");
+    ok(release instanceof AudioParam, "release is an AudioParam");
+    near(release.defaultValue, 0.25, "Correct default value for release");
+  }
+
+  compressor.threshold.value = -80;
+
+  osc.start();
+  var iteration = 0;
+  sp.onaudioprocess = function(e) {
+    if (iteration > 10) {
+      ok(compressor.reduction < 0,
+         "Feeding a full-scale sine to a compressor should result in an db" +
+         "reduction.");
+      sp.onaudioprocess = null;
+      osc.stop(0);
+
+      SimpleTest.finish();
+    }
+    iteration++;
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html b/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html
new file mode 100644
index 0000000000..1be838a4eb
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DynamicsCompressorNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var compressor = context.createDynamicsCompressor();
+
+    source.buffer = this.buffer;
+
+    source.connect(compressor);
+
+    var compressorWrapped = SpecialPowers.wrap(compressor);
+    ok("passThrough" in compressorWrapped, "DynamicsCompressorNode should support the passThrough API");
+    compressorWrapped.passThrough = true;
+
+    source.start(0);
+    return compressor;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html b/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html
new file mode 100644
index 0000000000..2e4b38ea50
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<meta charset="utf-8">
+  <title>Test DynamicsCompressor with Gain</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var samplerate = 44100;
+  var context = new OfflineAudioContext(1, samplerate/100, samplerate);
+
+  var osc = context.createOscillator();
+  osc.frequency.value = 2400;
+
+  var gain = context.createGain();
+  gain.gain.value = 1.5;
+
+  // These numbers are borrowed from the example code on MDN
+  // https://developer.mozilla.org/en-US/docs/Web/API/DynamicsCompressorNode
+  var compressor = context.createDynamicsCompressor();
+  compressor.threshold.value = -50;
+  compressor.knee.value = 40;
+  compressor.ratio.value = 12;
+  compressor.reduction.value = -20;
+  compressor.attack.value = 0;
+  compressor.release.value = 0.25;
+
+  osc.connect(gain);
+  gain.connect(compressor);
+  compressor.connect(context.destination);
+  osc.start();
+
+  context.startRendering().then(buffer => {
+    var peak = Math.max(...buffer.getChannelData(0));
+    console.log(peak);
+    // These values are experimentally determined. Without dynamics compression
+    // the peak should be just under 1.5. We also check for a minimum value
+    // to make sure we are not getting all zeros.
+    ok(peak >= 0.2 && peak < 1.0, "Peak value should be greater than 0.25 and less than 1.0");
+    SimpleTest.finish();
+  });
+});
+</script>
+<pre>
+</pre>
+</body>
diff --git a/dom/media/webaudio/test/test_gainNode.html b/dom/media/webaudio/test/test_gainNode.html
new file mode 100644
index 0000000000..41b19fda05
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNode.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    var gain = context.createGain();
+
+    source.buffer = buffer;
+
+    source.connect(gain);
+
+    ok(gain.gain, "The audioparam member must exist");
+    is(gain.gain.value, 1.0, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    gain.gain.value = 0.5;
+    is(gain.gain.value, 0.5, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    is(gain.channelCount, 2, "gain node has 2 input channels by default");
+    is(gain.channelCountMode, "max", "Correct channelCountMode for the gain node");
+    is(gain.channelInterpretation, "speakers", "Correct channelCountInterpretation for the gain node");
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate) / 2;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_gainNodeInLoop.html b/dom/media/webaudio/test/test_gainNodeInLoop.html
new file mode 100644
index 0000000000..6b32cbcfac
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNodeInLoop.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode in presence of loops</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+    source.loop = true;
+    source.start(0);
+    source.stop(sourceBuffer.duration * 2);
+
+    var gain = context.createGain();
+    // Adjust the gain in a way that we don't just end up modifying AudioChunk::mVolume
+    gain.gain.setValueAtTime(0.5, 0);
+    source.connect(gain);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 0; i < 4096; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 0.5;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_gainNodePassThrough.html b/dom/media/webaudio/test/test_gainNodePassThrough.html
new file mode 100644
index 0000000000..2a7cd6bf4c
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNodePassThrough.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var gain = context.createGain();
+
+    source.buffer = this.buffer;
+
+    source.connect(gain);
+
+    gain.gain.value = 0.5;
+
+    var gainWrapped = SpecialPowers.wrap(gain);
+    ok("passThrough" in gainWrapped, "GainNode should support the passThrough API");
+    gainWrapped.passThrough = true;
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_iirFilterNodePassThrough.html b/dom/media/webaudio/test/test_iirFilterNodePassThrough.html
new file mode 100644
index 0000000000..7773a5b827
--- /dev/null
+++ b/dom/media/webaudio/test/test_iirFilterNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var filter = context.createIIRFilter([0.5, 0.5], [1.0]);
+
+    source.buffer = this.buffer;
+
+    source.connect(filter);
+
+    var filterWrapped = SpecialPowers.wrap(filter);
+    ok("passThrough" in filterWrapped, "BiquadFilterNode should support the passThrough API");
+    filterWrapped.passThrough = true;
+
+    source.start(0);
+    return filter;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_maxChannelCount.html b/dom/media/webaudio/test/test_maxChannelCount.html
new file mode 100644
index 0000000000..319e2bf1e6
--- /dev/null
+++ b/dom/media/webaudio/test/test_maxChannelCount.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the AudioContext.destination interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// Work around bug 911777
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ok(ac.destination.maxChannelCount > 0, "We can query the maximum number of channels");
+
+  var oac = new OfflineAudioContext(2, 1024, 48000);
+  ok(oac.destination.maxChannelCount, 2, "This OfflineAudioContext should have 2 max channels.");
+
+  oac = new OfflineAudioContext(6, 1024, 48000);
+  ok(oac.destination.maxChannelCount, 6, "This OfflineAudioContext should have 6 max channels.");
+
+  expectException(function() {
+    oac.destination.channelCount = oac.destination.channelCount + 1;
+  }, DOMException.INDEX_SIZE_ERR);
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaDecoding.html b/dom/media/webaudio/test/test_mediaDecoding.html
new file mode 100644
index 0000000000..07e18162bc
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaDecoding.html
@@ -0,0 +1,367 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the decodeAudioData API and Resampling</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script type="text/javascript">
+
+// These routines have been copied verbatim from WebKit, and are used in order
+// to convert a memory buffer into a wave buffer.
+function writeString(s, a, offset) {
+    for (var i = 0; i < s.length; ++i) {
+        a[offset + i] = s.charCodeAt(i);
+    }
+}
+
+function writeInt16(n, a, offset) {
+    n = Math.floor(n);
+
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+}
+
+function writeInt32(n, a, offset) {
+    n = Math.floor(n);
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+    var b3 = (n >> 16) & 255;
+    var b4 = (n >> 24) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+    a[offset + 2] = b3;
+    a[offset + 3] = b4;
+}
+
+function writeAudioBuffer(audioBuffer, a, offset) {
+    var n = audioBuffer.length;
+    var channels = audioBuffer.numberOfChannels;
+
+    for (var i = 0; i < n; ++i) {
+        for (var k = 0; k < channels; ++k) {
+            var buffer = audioBuffer.getChannelData(k);
+            var sample = buffer[i] * 32768.0;
+
+            // Clip samples to the limitations of 16-bit.
+            // If we don't do this then we'll get nasty wrap-around distortion.
+            if (sample < -32768)
+                sample = -32768;
+            if (sample > 32767)
+                sample = 32767;
+
+            writeInt16(sample, a, offset);
+            offset += 2;
+        }
+    }
+}
+
+function createWaveFileData(audioBuffer) {
+    var frameLength = audioBuffer.length;
+    var numberOfChannels = audioBuffer.numberOfChannels;
+    var sampleRate = audioBuffer.sampleRate;
+    var bitsPerSample = 16;
+    var byteRate = sampleRate * numberOfChannels * bitsPerSample/8;
+    var blockAlign = numberOfChannels * bitsPerSample/8;
+    var wavDataByteLength = frameLength * numberOfChannels * 2; // 16-bit audio
+    var headerByteLength = 44;
+    var totalLength = headerByteLength + wavDataByteLength;
+
+    var waveFileData = new Uint8Array(totalLength);
+
+    var subChunk1Size = 16; // for linear PCM
+    var subChunk2Size = wavDataByteLength;
+    var chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
+
+    writeString("RIFF", waveFileData, 0);
+    writeInt32(chunkSize, waveFileData, 4);
+    writeString("WAVE", waveFileData, 8);
+    writeString("fmt ", waveFileData, 12);
+
+    writeInt32(subChunk1Size, waveFileData, 16);      // SubChunk1Size (4)
+    writeInt16(1, waveFileData, 20);                  // AudioFormat (2)
+    writeInt16(numberOfChannels, waveFileData, 22);   // NumChannels (2)
+    writeInt32(sampleRate, waveFileData, 24);         // SampleRate (4)
+    writeInt32(byteRate, waveFileData, 28);           // ByteRate (4)
+    writeInt16(blockAlign, waveFileData, 32);         // BlockAlign (2)
+    writeInt32(bitsPerSample, waveFileData, 34);      // BitsPerSample (4)
+
+    writeString("data", waveFileData, 36);
+    writeInt32(subChunk2Size, waveFileData, 40);      // SubChunk2Size (4)
+
+    // Write actual audio data starting at offset 44.
+    writeAudioBuffer(audioBuffer, waveFileData, 44);
+
+    return waveFileData;
+}
+
+</script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// fuzzTolerance and fuzzToleranceMobile are used to determine fuzziness
+// thresholds.  They're needed to make sure that we can deal with neglibible
+// differences in the binary buffer caused as a result of resampling the
+// audio.  fuzzToleranceMobile is typically larger on mobile platforms since
+// we do fixed-point resampling as opposed to floating-point resampling on
+// those platforms.
+var files = [
+  // An ogg file, 44.1khz, mono
+  {
+    url: "ting-44.1k-1ch.ogg",
+    valid: true,
+    expectedUrl: "ting-44.1k-1ch.wav",
+    numberOfChannels: 1,
+    frames: 30592,
+    sampleRate: 44100,
+    duration: 0.693,
+    fuzzTolerance: 5,
+    fuzzToleranceMobile: 1284
+  },
+  // An ogg file, 44.1khz, stereo
+  {
+    url: "ting-44.1k-2ch.ogg",
+    valid: true,
+    expectedUrl: "ting-44.1k-2ch.wav",
+    numberOfChannels: 2,
+    frames: 30592,
+    sampleRate: 44100,
+    duration: 0.693,
+    fuzzTolerance: 6,
+    fuzzToleranceMobile: 2544
+  },
+  // An ogg file, 48khz, mono
+  {
+    url: "ting-48k-1ch.ogg",
+    valid: true,
+    expectedUrl: "ting-48k-1ch.wav",
+    numberOfChannels: 1,
+    frames: 33297,
+    sampleRate: 48000,
+    duration: 0.693,
+    fuzzTolerance: 5,
+    fuzzToleranceMobile: 1388
+  },
+  // An ogg file, 48khz, stereo
+  {
+    url: "ting-48k-2ch.ogg",
+    valid: true,
+    expectedUrl: "ting-48k-2ch.wav",
+    numberOfChannels: 2,
+    frames: 33297,
+    sampleRate: 48000,
+    duration: 0.693,
+    fuzzTolerance: 14,
+    fuzzToleranceMobile: 2752
+  },
+  // Make sure decoding a wave file results in the same buffer (for both the
+  // resampling and non-resampling cases)
+  {
+    url: "ting-44.1k-1ch.wav",
+    valid: true,
+    expectedUrl: "ting-44.1k-1ch.wav",
+    numberOfChannels: 1,
+    frames: 30592,
+    sampleRate: 44100,
+    duration: 0.693,
+    fuzzTolerance: 0,
+    fuzzToleranceMobile: 0
+  },
+  {
+    url: "ting-48k-1ch.wav",
+    valid: true,
+    expectedUrl: "ting-48k-1ch.wav",
+    numberOfChannels: 1,
+    frames: 33297,
+    sampleRate: 48000,
+    duration: 0.693,
+    fuzzTolerance: 0,
+    fuzzToleranceMobile: 0
+  },
+  //  // A wave file
+  //  //{ url: "24bit-44khz.wav", valid: true, expectedUrl: "24bit-44khz-expected.wav" },
+  // A non-audio file
+  { url: "invalid.txt", valid: false, sampleRate: 44100 },
+  // A webm file with no audio
+  { url: "noaudio.webm", valid: false, sampleRate: 48000 },
+  // A video ogg file with audio
+  {
+    url: "audio.ogv",
+    valid: true,
+    expectedUrl: "audio-expected.wav",
+    numberOfChannels: 2,
+    sampleRate: 44100,
+    frames: 47680,
+    duration: 1.0807,
+    fuzzTolerance: 106,
+    fuzzToleranceMobile: 3482
+  }
+];
+
+// Returns true if the memory buffers are less different that |fuzz| bytes
+function fuzzyMemcmp(buf1, buf2, fuzz) {
+  var result = true;
+  var difference = 0;
+  is(buf1.length, buf2.length, "same length");
+  for (var i = 0; i < buf1.length; ++i) {
+    if (Math.abs(buf1[i] - buf2[i])) {
+      ++difference;
+    }
+  }
+  if (difference > fuzz) {
+    ok(false, "Expected at most " + fuzz + " bytes difference, found " + difference + " bytes");
+  }
+  return difference <= fuzz;
+}
+
+function getFuzzTolerance(test) {
+  var kIsMobile =
+    navigator.userAgent.indexOf("Mobile") != -1 || // b2g
+    navigator.userAgent.indexOf("Android") != -1;  // android
+  return kIsMobile ? test.fuzzToleranceMobile : test.fuzzTolerance;
+}
+
+function bufferIsSilent(buffer) {
+  for (var i = 0; i < buffer.length; ++i) {
+    if (buffer.getChannelData(0)[i] != 0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+function checkAudioBuffer(buffer, test) {
+  if (buffer.numberOfChannels != test.numberOfChannels) {
+    is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
+    return;
+  }
+  ok(Math.abs(buffer.duration - test.duration) < 1e-3, "Correct duration");
+  if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
+    ok(false, "got: " + buffer.duration  + ", expected: " + test.duration);
+  }
+  is(buffer.sampleRate, test.sampleRate, "Correct sample rate");
+  is(buffer.length, test.frames, "Correct length");
+
+  var wave = createWaveFileData(buffer);
+  ok(fuzzyMemcmp(wave, test.expectedWaveData, getFuzzTolerance(test)), "Received expected decoded data");
+}
+
+function checkResampledBuffer(buffer, test, callback) {
+  if (buffer.numberOfChannels != test.numberOfChannels) {
+    is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
+    return;
+  }
+  ok(Math.abs(buffer.duration - test.duration) < 1e-3, "Correct duration");
+  if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
+    ok(false, "got: " + buffer.duration  + ", expected: " + test.duration);
+  }
+  // Take into account the resampling when checking the size
+  var expectedLength = test.frames * buffer.sampleRate / test.sampleRate;
+  ok(Math.abs(buffer.length - expectedLength) < 1.0, "Correct length", "got " + buffer.length + ", expected about " + expectedLength);
+
+  // Playback the buffer in the original context, to resample back to the
+  // original rate and compare with the decoded buffer without resampling.
+  cx = test.nativeContext;
+  var expected = cx.createBufferSource();
+  expected.buffer = test.expectedBuffer;
+  expected.start();
+  var inverse = cx.createGain();
+  inverse.gain.value = -1;
+  expected.connect(inverse);
+  inverse.connect(cx.destination);
+  var resampled = cx.createBufferSource();
+  resampled.buffer = buffer;
+  resampled.start();
+  // This stop should do nothing, but it tests for bug 937475
+  resampled.stop(test.frames / cx.sampleRate);
+  resampled.connect(cx.destination);
+  cx.oncomplete = function(e) {
+    ok(!bufferIsSilent(e.renderedBuffer), "Expect buffer not silent");
+    // Resampling will lose the highest frequency components, so we should
+    // pass the difference through a low pass filter.  However, either the
+    // input files don't have significant high frequency components or the
+    // tolerance in compareBuffers() is too high to detect them.
+    compareBuffers(e.renderedBuffer,
+                   cx.createBuffer(test.numberOfChannels,
+                                   test.frames, test.sampleRate));
+    callback();
+  }
+  cx.startRendering();
+}
+
+function runResampling(test, response, callback) {
+  var sampleRate = test.sampleRate == 44100 ? 48000 : 44100;
+  var cx = new OfflineAudioContext(1, 1, sampleRate);
+  cx.decodeAudioData(response, function onSuccess(asyncResult) {
+    is(asyncResult.sampleRate, sampleRate, "Correct sample rate");
+
+    checkResampledBuffer(asyncResult, test, callback);
+  }, function onFailure() {
+    ok(false, "Expected successful decode with resample");
+    callback();
+  });
+}
+
+function runTest(test, response, callback) {
+  // We need to copy the array here, because decodeAudioData will detach the
+  // array's buffer.
+  var compressedAudio = response.slice(0);
+  var expectCallback = false;
+  var cx = new OfflineAudioContext(test.numberOfChannels || 1,
+                                   test.frames || 1, test.sampleRate);
+  cx.decodeAudioData(response, function onSuccess(asyncResult) {
+    ok(expectCallback, "Success callback should fire asynchronously");
+    ok(test.valid, "Did expect success for test " + test.url);
+
+    checkAudioBuffer(asyncResult, test);
+
+    test.expectedBuffer = asyncResult;
+    test.nativeContext = cx;
+    runResampling(test, compressedAudio, callback);
+  }, function onFailure() {
+    ok(expectCallback, "Failure callback should fire asynchronously");
+    ok(!test.valid, "Did expect failure for test " + test.url);
+    callback();
+  });
+  expectCallback = true;
+}
+
+function loadTest(test, callback) {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", test.url, true);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    var getExpected = new XMLHttpRequest();
+    getExpected.open("GET", test.expectedUrl, true);
+    getExpected.responseType = "arraybuffer";
+    getExpected.onload = function() {
+      test.expectedWaveData = new Uint8Array(getExpected.response);
+      runTest(test, xhr.response, callback);
+    };
+    getExpected.send();
+  };
+  xhr.send();
+}
+
+function loadNextTest() {
+  if (files.length) {
+    loadTest(files.shift(), loadNextTest);
+  } else {
+    SimpleTest.finish();
+  }
+}
+
+loadNextTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html
new file mode 100644
index 0000000000..3e196735f7
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var expectedMinNonzeroSampleCount;
+  var expectedMaxNonzeroSampleCount;
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    is(e.inputBuffer.numberOfChannels, 1,
+       "Checking data channel count (nonzeroSamplesThisBuffer=" +
+       nonzeroSamplesThisBuffer + ")");
+    ok(nonzeroSampleCount <= expectedMaxNonzeroSampleCount,
+       "Too many nonzero samples (got " + nonzeroSampleCount + ", expected max " + expectedMaxNonzeroSampleCount + ")");
+    if (nonzeroSampleCount >= expectedMinNonzeroSampleCount &&
+        nonzeroSamplesThisBuffer == 0) {
+      ok(true,
+       "Check received enough nonzero samples (got " + nonzeroSampleCount + ", expected min " + expectedMinNonzeroSampleCount + ")");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.onloadedmetadata = function() {
+    var node = context.createMediaElementSource(audio);
+    var sp = context.createScriptProcessor(2048, 1);
+    node.connect(sp);
+    // Use a fuzz factor of 100 to account for samples that just happen to be zero
+    expectedMinNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) - 100;
+    expectedMaxNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) + 500;
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html
new file mode 100644
index 0000000000..7e03b7079c
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html
@@ -0,0 +1,94 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+// Turn off the authentication dialog blocking for this test.
+SpecialPowers.setIntPref("network.auth.subresource-http-auth-allow", 2)
+
+var tests = [
+  // Not the same origin no CORS asked for, should have silence
+  { url: "http://example.org:80/tests/dom/media/webaudio/test/small-shot.ogg",
+    cors: null,
+    expectSilence: true },
+  // Same origin, should have sound
+  { url: "small-shot.ogg",
+    cors: null,
+    expectSilence: false },
+  // Cross-origin but we asked for CORS and the server answered with the right
+  // header, should have
+  { url: "http://example.org:80/tests/dom/media/webaudio/test/corsServer.sjs",
+    cors: "anonymous",
+    expectSilence: false }
+];
+
+var testsRemaining = tests.length;
+
+tests.forEach(function(e) {
+  e.ac = new AudioContext();
+  var a = new Audio();
+  if (e.cors) {
+    a.crossOrigin = e.cors;
+  }
+  a.src = e.url;
+  document.body.appendChild(a);
+
+  a.onloadedmetadata = () => {
+    // Wait for "loadedmetadata" before capturing since tracks are then known
+    // directly. If we set up the capture before "loadedmetadata" we
+    // (internally) have to wait an extra async jump for tracks to become known
+    // to main thread, before setting up audio data forwarding to the node.
+    // As that happens, the audio resource may have already ended on slow test
+    // machines, causing failures.
+    a.onloadedmetadata = null;
+    var measn = e.ac.createMediaElementSource(a);
+    var sp = e.ac.createScriptProcessor(2048, 1);
+    sp.seenSound = false;
+    sp.onaudioprocess = checkBufferSilent;
+
+    measn.connect(sp);
+    a.play();
+  };
+
+  function checkFinished(sp) {
+    if (a.ended) {
+      sp.onaudioprocess = null;
+      var not = e.expectSilence ? "" : "not";
+      is(e.expectSilence, !sp.seenSound,
+         "Buffer is " + not + " silent as expected, for " +
+         e.url + " (cors: " + e.cors + ")");
+      if (--testsRemaining == 0) {
+        SimpleTest.finish();
+      }
+    }
+  }
+
+  function checkBufferSilent(e) {
+    var inputArrayBuffer = e.inputBuffer.getChannelData(0);
+    var silent = true;
+    for (var i = 0; i < inputArrayBuffer.length; i++) {
+      if (inputArrayBuffer[i] != 0.0) {
+        silent = false;
+        break;
+      }
+    }
+    // It is acceptable to find a full buffer of silence here, even if we expect
+    // sound, because Gecko's looping on media elements is not seamless and we
+    // can underrun. We are looking for at least one buffer of non-silent data.
+    e.target.seenSound = !silent || e.target.seenSound;
+    checkFinished(e.target);
+    return silent;
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html
new file mode 100644
index 0000000000..8d3b0ed465
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html
@@ -0,0 +1,128 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function binIndexForFrequency(frequency, analyser) {
+  return 1 + Math.round(frequency *
+                        analyser.fftSize /
+                        analyser.context.sampleRate);
+}
+
+function debugCanvas(analyser) {
+  var cvs = document.createElement("canvas");
+  document.body.appendChild(cvs);
+
+  // Easy: 1px per bin
+  cvs.width = analyser.frequencyBinCount;
+  cvs.height = 256;
+  cvs.style.border = "1px solid red";
+
+  var c = cvs.getContext('2d');
+  var buf = new Uint8Array(analyser.frequencyBinCount);
+
+  function render() {
+    c.clearRect(0, 0, cvs.width, cvs.height);
+    analyser.getByteFrequencyData(buf);
+    for (var i = 0; i < buf.length; i++) {
+      c.fillRect(i, (256 - (buf[i])), 1, 256);
+    }
+    requestAnimationFrame(render);
+  }
+  requestAnimationFrame(render);
+}
+
+
+function checkFrequency(an) {
+  an.getFloatFrequencyData(frequencyArray);
+  // We should have no energy when checking the data largely outside the index
+  // for 440Hz (the frequency of the sine wave), start checking an octave above,
+  // the Opus compression can add some harmonics to the pure since wave.
+  var index = binIndexForFrequency(880, an);
+  var underTreshold = true;
+  for (var i = index; i < frequencyArray.length; i++) {
+    // Let some slack, there might be some noise here because of int -> float
+    // conversion or the Opus encoding.
+    if (frequencyArray[i] > an.minDecibels + 40) {
+      return false;
+    }
+  }
+
+  // On the other hand, we should find a peak at 440Hz. Our sine wave is not
+  // attenuated, we're expecting the peak to reach 0dBFs.
+  index = binIndexForFrequency(440, an);
+  info("energy at 440: " + frequencyArray[index] + ", threshold " + (an.maxDecibels - 10));
+  if (frequencyArray[index] < (an.maxDecibels - 10)) {
+    return false;
+  }
+
+  return true;
+}
+
+var audioElement = new Audio();
+audioElement.src = 'sine-440-10s.opus'
+audioElement.loop = true;
+var ac = new AudioContext();
+var mediaElementSource = ac.createMediaElementSource(audioElement);
+var an = ac.createAnalyser();
+frequencyArray = new Float32Array(an.frequencyBinCount);
+
+// Uncomment this to check what the analyser is doing.
+// debugCanvas(an);
+
+mediaElementSource.connect(an)
+
+audioElement.play();
+// We want to check the we have the expected audio for at least two loop of
+// the HTMLMediaElement, piped into an AudioContext. The file is ten seconds,
+// and we use the default FFT size.
+var lastCurrentTime = 0;
+var loopCount = 0;
+audioElement.onplaying = function() {
+  audioElement.ontimeupdate = function() {
+    // We don't run the analysis when close to loop point or at the
+    // beginning, since looping is not seamless, there could be an
+    // unpredictable amount of silence
+    var rv = checkFrequency(an);
+    info("currentTime: " + audioElement.currentTime);
+    if (audioElement.currentTime < 4 ||
+        audioElement.currentTIme > 8){
+      return;
+    }
+    if (!rv) {
+      ok(false, "Found unexpected noise during analysis.");
+      audioElement.ontimeupdate = null;
+      audioElement.onplaying = null;
+      ac.close();
+      audioElement.src = '';
+      SimpleTest.finish()
+      return;
+    }
+    ok(true, "Found correct audio signal during analysis");
+    info(lastCurrentTime + " " + audioElement.currentTime);
+    if (lastCurrentTime > audioElement.currentTime) {
+      info("loopCount: " + loopCount);
+      if (loopCount > 1) {
+        audioElement.ontimeupdate = null;
+        audioElement.onplaying = null;
+        ac.close();
+        audioElement.src = '';
+        SimpleTest.finish();
+      }
+      lastCurrentTime = audioElement.currentTime;
+      loopCount++;
+    } else {
+      lastCurrentTime = audioElement.currentTime;
+    }
+  }
+}
+
+</script>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html
new file mode 100644
index 0000000000..1bb0ad9ec7
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var node = context.createMediaElementSource(audio);
+  var sp = context.createScriptProcessor(2048, 1);
+  node.connect(sp);
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  var srcWrapped = SpecialPowers.wrap(node);
+  ok("passThrough" in srcWrapped, "MediaElementAudioSourceNode should support the passThrough API");
+  srcWrapped.passThrough = true;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    if (iterationCount == 10) {
+      is(nonzeroSampleCount, 0, "The input must be silence");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.oncanplaythrough = function() {
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html
new file mode 100644
index 0000000000..ad0b355b18
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode before "loadedmetadata"</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+var video = document.createElement("video");
+function test() {
+  video.src = "audiovideo.mp4";
+
+  var context = new AudioContext();
+  var complete = false;
+
+  video.onended = () => {
+    if (complete) {
+      return;
+    }
+
+    complete = true;
+    ok(false, "Video ended without any samples seen");
+    SimpleTest.finish();
+  };
+
+  video.ontimeupdate = () => {
+    info("Timeupdate: " + video.currentTime);
+  };
+
+  var node = context.createMediaElementSource(video);
+  var sp = context.createScriptProcessor(2048, 1);
+  node.connect(sp);
+
+  // This test ensures we receive some nonzero samples when we capture to
+  // WebAudio before "loadedmetadata".
+  sp.onaudioprocess = e => {
+    if (complete) {
+      return;
+    }
+
+    var buf = e.inputBuffer.getChannelData(0);
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        complete = true;
+        ok(true, "Got non-zero samples");
+        SimpleTest.finish();
+        return;
+      }
+    }
+  };
+
+  video.play();
+}
+
+if (video.canPlayType("video/mp4")) {
+  test();
+} else {
+  ok(true, "MP4 not supported. Skipping.");
+  SimpleTest.finish();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html b/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html
new file mode 100644
index 0000000000..5aa1a79105
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test MediaStreamAudioDestinationNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<audio id="audioelem"></audio>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test uses a live media element so it needs to wait for the media stack to do some work.");
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+
+  var dest = context.createMediaStreamDestination();
+  source.connect(dest);
+
+  var elem = document.getElementById('audioelem');
+  elem.srcObject = dest.stream;
+  elem.onloadedmetadata = function() {
+    ok(true, "got metadata event");
+    setTimeout(function() {
+      is(elem.played.length, 1, "should have a played interval");
+      is(elem.played.start(0), 0, "should have played immediately");
+      isnot(elem.played.end(0), 0, "should have played for a non-zero interval");
+
+      // This will end the media element.
+      dest.stream.getTracks()[0].stop();
+    }, 2000);
+  };
+  elem.onended = function() {
+    ok(true, "media element ended after destination track.stop()");
+    SimpleTest.finish();
+  };
+
+  source.start(0);
+  elem.play();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html
new file mode 100644
index 0000000000..85d96d3e8a
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode processing is correct</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createBuffer(context) {
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    buffer.getChannelData(1)[i] = -buffer.getChannelData(0)[i];
+  }
+  return buffer;
+}
+
+var gTest = {
+  length: 2048,
+  skipOfflineContextTests: true,
+  createGraph: function(context) {
+    var sourceGraph = new AudioContext();
+    var source = sourceGraph.createBufferSource();
+    source.buffer = createBuffer(context);
+    var dest = sourceGraph.createMediaStreamDestination();
+    source.connect(dest);
+    source.start(0);
+
+    var mediaStreamSource = context.createMediaStreamSource(dest.stream);
+    // channelCount and channelCountMode should have no effect
+    mediaStreamSource.channelCount = 1;
+    mediaStreamSource.channelCountMode = "explicit";
+    return mediaStreamSource;
+  },
+  createExpectedBuffers: function(context) {
+    return createBuffer(context);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html
new file mode 100644
index 0000000000..f3cc0334a2
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+var audio = new Audio("http://example.org:80/tests/dom/media/webaudio/test/small-shot.ogg");
+var context = new AudioContext();
+var node = context.createMediaStreamSource(audio.mozCaptureStreamUntilEnded());
+var sp = context.createScriptProcessor(2048, 1);
+node.connect(sp);
+var nonzeroSampleCount = 0;
+var complete = false;
+var iterationCount = 0;
+
+// This test ensures we receive at least expectedSampleCount nonzero samples
+function processSamples(e) {
+  if (complete) {
+    return;
+  }
+
+  if (iterationCount == 0) {
+    // Don't start playing the audio until the AudioContext stuff is connected
+    // and running.
+    audio.play();
+  }
+  ++iterationCount;
+
+  var buf = e.inputBuffer.getChannelData(0);
+  var nonzeroSamplesThisBuffer = 0;
+  for (var i = 0; i < buf.length; ++i) {
+    if (buf[i] != 0) {
+      ++nonzeroSamplesThisBuffer;
+    }
+  }
+  is(nonzeroSamplesThisBuffer, 0,
+     "Checking all samples are zero");
+  if (iterationCount >= 20) {
+    SimpleTest.finish();
+    complete = true;
+  }
+}
+
+audio.oncanplaythrough = function() {
+  sp.onaudioprocess = processSamples;
+};
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html
new file mode 100644
index 0000000000..7a9b6c4a61
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html
@@ -0,0 +1,89 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test that MediaStreamAudioSourceNode and its input MediaStream stays alive while there are active tracks</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("gUM and WebAudio data is async to main thread. " +
+                               "We need a timeout to see that something does " +
+                               "NOT happen to data.");
+
+var context = new AudioContext();
+var analyser = context.createAnalyser();
+
+function wait(millis, resolveWithThis) {
+  return new Promise(resolve => setTimeout(() => resolve(resolveWithThis), millis));
+}
+
+function binIndexForFrequency(frequency) {
+  return 1 + Math.round(frequency * analyser.fftSize / context.sampleRate);
+}
+
+function waitForAudio(analysisFunction, cancelPromise) {
+  var data = new Uint8Array(analyser.frequencyBinCount);
+  var cancelled = false;
+  var cancelledMsg = "";
+  cancelPromise.then(msg => {
+    cancelled = true;
+    cancelledMsg = msg;
+  });
+  return new Promise((resolve, reject) => {
+    var loop = () => {
+      analyser.getByteFrequencyData(data);
+      if (cancelled) {
+        reject(new Error("waitForAudio cancelled: " + cancelledMsg));
+        return;
+      }
+      if (analysisFunction(data)) {
+        resolve();
+        return;
+      }
+      requestAnimationFrame(loop);
+    };
+    loop();
+  });
+}
+
+navigator.mediaDevices.getUserMedia({audio: true, fake: true})
+  .then(stream => {
+    stream.onended = () => ended = true;
+    let source = context.createMediaStreamSource(stream);
+    source.connect(analyser);
+    analyser.connect(context.destination);
+  })
+  .then(() => {
+    ok(true, "Waiting for audio to pass through the analyser")
+    return waitForAudio(arr => arr[binIndexForFrequency(1000)] > 200,
+                        wait(60000, "Timeout waiting for audio"));
+  })
+  .then(() => {
+    ok(true, "Audio was detected by the analyser. Forcing CC.");
+    SpecialPowers.forceCC();
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    SpecialPowers.forceGC();
+
+    info("Checking that GC didn't destroy the stream or source node");
+    return waitForAudio(arr => arr[binIndexForFrequency(1000)] < 50,
+                        wait(5000, "Timeout waiting for GC (timeout OK)"))
+                  .then(() => Promise.reject("Audio stopped unexpectedly"),
+                        () => Promise.resolve());
+  })
+  .then(() => {
+    ok(true, "Audio is still flowing");
+    SimpleTest.finish();
+  })
+  .catch(e => {
+    ok(false, "Error executing test: " + e + (e.stack ? "\n" + e.stack : ""));
+    SimpleTest.finish();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html
new file mode 100644
index 0000000000..d2c22600a7
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createBuffer(context, delay) {
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  for (var i = 0; i < 2048 - delay; ++i) {
+    buffer.getChannelData(0)[i + delay] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    buffer.getChannelData(1)[i + delay] = -buffer.getChannelData(0)[i + delay];
+  }
+  return buffer;
+}
+
+var gTest = {
+  length: 2048,
+  skipOfflineContextTests: true,
+  createGraph: function(context) {
+    var sourceGraph = new AudioContext();
+    var source = sourceGraph.createBufferSource();
+    source.buffer = createBuffer(context, 0);
+    var dest = sourceGraph.createMediaStreamDestination();
+    source.connect(dest);
+    source.start(0);
+
+    var mediaStreamSource = context.createMediaStreamSource(dest.stream);
+    // channelCount and channelCountMode should have no effect
+    mediaStreamSource.channelCount = 1;
+    mediaStreamSource.channelCountMode = "explicit";
+
+    var srcWrapped = SpecialPowers.wrap(mediaStreamSource);
+    ok("passThrough" in srcWrapped, "MediaStreamAudioSourceNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    return mediaStreamSource;
+  },
+  createExpectedBuffers: function(context) {
+    return context.createBuffer(2, 2048, context.sampleRate);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html
new file mode 100644
index 0000000000..4a4f03c53b
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode processing is correct</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var expectedMinNonzeroSampleCount;
+  var expectedMaxNonzeroSampleCount;
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    is(e.inputBuffer.numberOfChannels, 1,
+       "Checking data channel count (nonzeroSamplesThisBuffer=" +
+       nonzeroSamplesThisBuffer + ")");
+    ok(nonzeroSampleCount <= expectedMaxNonzeroSampleCount,
+       "Too many nonzero samples (got " + nonzeroSampleCount + ", expected max " + expectedMaxNonzeroSampleCount + ")");
+    if (nonzeroSampleCount >= expectedMinNonzeroSampleCount &&
+        nonzeroSamplesThisBuffer == 0) {
+      ok(true,
+       "Check received enough nonzero samples (got " + nonzeroSampleCount + ", expected min " + expectedMinNonzeroSampleCount + ")");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.onloadedmetadata = function() {
+    var node = context.createMediaStreamSource(audio.mozCaptureStreamUntilEnded());
+    var sp = context.createScriptProcessor(2048, 1, 0);
+    node.connect(sp);
+    // Use a fuzz factor of 100 to account for samples that just happen to be zero
+    expectedMinNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) - 100;
+    expectedMaxNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) + 500;
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+  
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mixingRules.html b/dom/media/webaudio/test/test_mixingRules.html
new file mode 100644
index 0000000000..0bdcff87ef
--- /dev/null
+++ b/dom/media/webaudio/test/test_mixingRules.html
@@ -0,0 +1,401 @@
+<!DOCTYPE html>
+<html>
+<head>
+  <title>Testcase for AudioNode channel up-mix/down-mix rules</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+
+<body>
+
+<script>
+
+// This test is based on http://src.chromium.org/viewvc/blink/trunk/LayoutTests/webaudio/audionode-channel-rules.html
+
+var context = null;
+var sp = null;
+var renderNumberOfChannels = 8;
+var singleTestFrameLength = 8;
+var testBuffers;
+
+// A list of connections to an AudioNode input, each of which is to be used in one or more specific test cases.
+// Each element in the list is a string, with the number of connections corresponding to the length of the string,
+// and each character in the string is from '1' to '8' representing a 1 to 8 channel connection (from an AudioNode output).
+// For example, the string "128" means 3 connections, having 1, 2, and 8 channels respectively.
+var connectionsList = [];
+for (var i = 1; i <= 8; ++i) {
+  connectionsList.push(i.toString());
+  for (var j = 1; j <= 8; ++j) {
+    connectionsList.push(i.toString() + j.toString());
+  }
+}
+
+// A list of mixing rules, each of which will be tested against all of the connections in connectionsList.
+var mixingRulesList = [
+    {channelCount: 1, channelCountMode: "max", channelInterpretation: "speakers"},
+    {channelCount: 2, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 3, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 4, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 5, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 6, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 7, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 2, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 3, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 4, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 5, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 6, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 7, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 8, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 1, channelCountMode: "max", channelInterpretation: "discrete"},
+    {channelCount: 2, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 3, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 4, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 5, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 6, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 3, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 4, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 5, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 6, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 7, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 8, channelCountMode: "explicit", channelInterpretation: "discrete"},
+];
+
+var numberOfTests = mixingRulesList.length * connectionsList.length;
+
+// Create an n-channel buffer, with all sample data zero except for a shifted impulse.
+// The impulse position depends on the channel index.
+// For example, for a 4-channel buffer:
+// channel0: 1 0 0 0 0 0 0 0
+// channel1: 0 1 0 0 0 0 0 0
+// channel2: 0 0 1 0 0 0 0 0
+// channel3: 0 0 0 1 0 0 0 0
+function createTestBuffer(numberOfChannels) {
+    var buffer = context.createBuffer(numberOfChannels, singleTestFrameLength, context.sampleRate);
+    for (var i = 0; i < numberOfChannels; ++i) {
+        var data = buffer.getChannelData(i);
+        data[i] = 1;
+    }
+    return buffer;
+}
+
+// Discrete channel interpretation mixing:
+// https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html#UpMix
+// up-mix by filling channels until they run out then ignore remaining dest channels.
+// down-mix by filling as many channels as possible, then dropping remaining source channels.
+function discreteSum(sourceBuffer, destBuffer) {
+    if (sourceBuffer.length != destBuffer.length) {
+        is(sourceBuffer.length, destBuffer.length, "source and destination buffers should have the same length");
+    }
+
+    var numberOfChannels = Math.min(sourceBuffer.numberOfChannels, destBuffer.numberOfChannels);
+    var length = sourceBuffer.length;
+
+    for (var c = 0; c < numberOfChannels; ++c) {
+        var source = sourceBuffer.getChannelData(c);
+        var dest = destBuffer.getChannelData(c);
+        for (var i = 0; i < length; ++i) {
+            dest[i] += source[i];
+        }
+    }
+}
+
+// Speaker channel interpretation mixing:
+// https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html#UpMix
+function speakersSum(sourceBuffer, destBuffer)
+{
+    var numberOfSourceChannels = sourceBuffer.numberOfChannels;
+    var numberOfDestinationChannels = destBuffer.numberOfChannels;
+    var length = destBuffer.length;
+
+    if ((numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) ||
+        (numberOfDestinationChannels == 4 && numberOfSourceChannels == 1)) {
+        // Handle mono -> stereo/Quad case (summing mono channel into both left and right).
+        var source = sourceBuffer.getChannelData(0);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += source[i];
+            destR[i] += source[i];
+        }
+        } else if ((numberOfDestinationChannels == 4 && numberOfSourceChannels == 2) ||
+                   (numberOfDestinationChannels == 6 && numberOfSourceChannels == 2)) {
+        // Handle stereo -> Quad/5.1 case (summing left and right channels into the output's left and right).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i];
+            destR[i] += sourceR[i];
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) {
+        // Handle stereo -> mono case. output += 0.5 * (input.L + input.R).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.5 * (sourceL[i] + sourceR[i]);
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 4) {
+        // Handle Quad -> mono case. output += 0.25 * (input.L + input.R + input.SL + input.SR).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.25 * (sourceL[i] + sourceR[i] + sourceSL[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 4) {
+        // Handle Quad -> stereo case. outputLeft += 0.5 * (input.L + input.SL),
+        //                             outputRight += 0.5 * (input.R + input.SR).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += 0.5 * (sourceL[i] + sourceSL[i]);
+            destR[i] += 0.5 * (sourceR[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 4) {
+        // Handle Quad -> 5.1 case. outputLeft += (inputL, inputR, 0, 0, inputSL, inputSR)
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+        var destSL = destBuffer.getChannelData(4);
+        var destSR = destBuffer.getChannelData(5);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i];
+            destR[i] += sourceR[i];
+            destSL[i] += sourceSL[i];
+            destSR[i] += sourceSR[i];
+        }
+    } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) {
+        // Handle mono -> 5.1 case, sum mono channel into center.
+        var source = sourceBuffer.getChannelData(0);
+        var dest = destBuffer.getChannelData(2);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += source[i];
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> mono.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.7071 * (sourceL[i] + sourceR[i]) + sourceC[i] + 0.5 * (sourceSL[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> stereo.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i] + 0.7071 * (sourceC[i] + sourceSL[i]);
+            destR[i] += sourceR[i] + 0.7071 * (sourceC[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 4 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> Quad.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+        var destSL = destBuffer.getChannelData(2);
+        var destSR = destBuffer.getChannelData(3);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i] + 0.7071 * sourceC[i];
+            destR[i] += sourceR[i] + 0.7071 * sourceC[i];
+            destSL[i] += sourceSL[i];
+            destSR[i] += sourceSR[i];
+        }
+    } else {
+        // Fallback for unknown combinations.
+        discreteSum(sourceBuffer, destBuffer);
+    }
+}
+
+function scheduleTest(testNumber, connections, channelCount, channelCountMode, channelInterpretation) {
+    var mixNode = context.createGain();
+    mixNode.channelCount = channelCount;
+    mixNode.channelCountMode = channelCountMode;
+    mixNode.channelInterpretation = channelInterpretation;
+    mixNode.connect(sp);
+
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+
+        var source = context.createBufferSource();
+        // Get a buffer with the right number of channels, converting from 1-based to 0-based index.
+        var buffer = testBuffers[connectionNumberOfChannels - 1];
+        source.buffer = buffer;
+        source.connect(mixNode);
+
+        // Start at the right offset.
+        var sampleFrameOffset = testNumber * singleTestFrameLength;
+        var time = sampleFrameOffset / context.sampleRate;
+        source.start(time);
+    }
+}
+
+function computeNumberOfChannels(connections, channelCount, channelCountMode) {
+    if (channelCountMode == "explicit")
+        return channelCount;
+
+    var computedNumberOfChannels = 1; // Must have at least one channel.
+
+    // Compute "computedNumberOfChannels" based on all the connections.
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+        computedNumberOfChannels = Math.max(computedNumberOfChannels, connectionNumberOfChannels);
+    }
+
+    if (channelCountMode == "clamped-max")
+        computedNumberOfChannels = Math.min(computedNumberOfChannels, channelCount);
+
+    return computedNumberOfChannels;
+}
+
+function checkTestResult(renderedBuffer, testNumber, connections, channelCount, channelCountMode, channelInterpretation) {
+    var computedNumberOfChannels = computeNumberOfChannels(connections, channelCount, channelCountMode);
+
+    // Create a zero-initialized silent AudioBuffer with computedNumberOfChannels.
+    var destBuffer = context.createBuffer(computedNumberOfChannels, singleTestFrameLength, context.sampleRate);
+
+    // Mix all of the connections into the destination buffer.
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+        var sourceBuffer = testBuffers[connectionNumberOfChannels - 1]; // convert from 1-based to 0-based index
+
+        if (channelInterpretation == "speakers") {
+            speakersSum(sourceBuffer, destBuffer);
+        } else if (channelInterpretation == "discrete") {
+            discreteSum(sourceBuffer, destBuffer);
+        } else {
+            ok(false, "Invalid channel interpretation!");
+        }
+    }
+
+    // Validate that destBuffer matches the rendered output.
+    // We need to check the rendered output at a specific sample-frame-offset corresponding
+    // to the specific test case we're checking for based on testNumber.
+
+    var sampleFrameOffset = testNumber * singleTestFrameLength;
+    for (var c = 0; c < renderNumberOfChannels; ++c) {
+        var renderedData = renderedBuffer.getChannelData(c);
+        for (var frame = 0; frame < singleTestFrameLength; ++frame) {
+            var renderedValue = renderedData[frame + sampleFrameOffset];
+
+            var expectedValue = 0;
+            if (c < destBuffer.numberOfChannels) {
+                var expectedData = destBuffer.getChannelData(c);
+                expectedValue = expectedData[frame];
+            }
+
+            if (Math.abs(renderedValue - expectedValue) > 1e-4) {
+                var s = "connections: " + connections + ", " + channelCountMode;
+
+                // channelCount is ignored in "max" mode.
+                if (channelCountMode == "clamped-max" || channelCountMode == "explicit") {
+                    s += "(" + channelCount + ")";
+                }
+
+                s += ", " + channelInterpretation + ". ";
+
+                var message = s + "rendered: " + renderedValue + " expected: " + expectedValue + " channel: " + c + " frame: " + frame;
+                is(renderedValue, expectedValue, message);
+            }
+        }
+    }
+}
+
+function checkResult(event) {
+    var buffer = event.inputBuffer;
+
+    // Sanity check result.
+    ok(buffer.length != numberOfTests * singleTestFrameLength ||
+       buffer.numberOfChannels != renderNumberOfChannels, "Sanity check");
+
+    // Check all the tests.
+    var testNumber = 0;
+    for (var m = 0; m < mixingRulesList.length; ++m) {
+        var mixingRules = mixingRulesList[m];
+        for (var i = 0; i < connectionsList.length; ++i, ++testNumber) {
+            checkTestResult(buffer, testNumber, connectionsList[i], mixingRules.channelCount, mixingRules.channelCountMode, mixingRules.channelInterpretation);
+        }
+    }
+
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+}
+
+SimpleTest.waitForExplicitFinish();
+function runTest() {
+    // Create 8-channel offline audio context.
+    // Each test will render 8 sample-frames starting at sample-frame position testNumber * 8.
+    var totalFrameLength = numberOfTests * singleTestFrameLength;
+    context = new AudioContext();
+    var nextPowerOfTwo = 256;
+    while (nextPowerOfTwo < totalFrameLength) {
+        nextPowerOfTwo *= 2;
+    }
+    sp = context.createScriptProcessor(nextPowerOfTwo, renderNumberOfChannels);
+
+    // Set destination to discrete mixing.
+    sp.channelCount = renderNumberOfChannels;
+    sp.channelCountMode = "explicit";
+    sp.channelInterpretation = "discrete";
+
+    // Create test buffers from 1 to 8 channels.
+    testBuffers = new Array();
+    for (var i = 0; i < renderNumberOfChannels; ++i) {
+        testBuffers[i] = createTestBuffer(i + 1);
+    }
+
+    // Schedule all the tests.
+    var testNumber = 0;
+    for (var m = 0; m < mixingRulesList.length; ++m) {
+        var mixingRules = mixingRulesList[m];
+        for (var i = 0; i < connectionsList.length; ++i, ++testNumber) {
+            scheduleTest(testNumber, connectionsList[i], mixingRules.channelCount, mixingRules.channelCountMode, mixingRules.channelInterpretation);
+        }
+    }
+
+    // Render then check results.
+    sp.onaudioprocess = checkResult;
+}
+
+runTest();
+
+</script>
+
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mozaudiochannel.html b/dom/media/webaudio/test/test_mozaudiochannel.html
new file mode 100644
index 0000000000..6ba14347bf
--- /dev/null
+++ b/dom/media/webaudio/test/test_mozaudiochannel.html
@@ -0,0 +1,151 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test for mozaudiochannel</title>
+  <script type="application/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="/tests/SimpleTest/EventUtils.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css"/>
+</head>
+<body>
+<p id="display"></p>
+<pre id="test">
+<script type="application/javascript">
+
+function test_basic() {
+  var ac = new AudioContext();
+  ok(ac, "AudioContext created");
+
+  // Default
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  // Unpermitted channels
+  ac = new AudioContext("content");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  ac = new AudioContext("notification");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  ac = new AudioContext("alarm");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  ac = new AudioContext("telephony");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  ac = new AudioContext("ringer");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  ac = new AudioContext("publicnotification");
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  runTest();
+}
+
+function test_permission(aChannel) {
+  var ac = new AudioContext();
+  ok(ac, "AudioContext created");
+
+  is(ac.mozAudioChannelType, "normal", "Default ac channel == 'normal'");
+
+  var channel = SpecialPowers.wrap(ac).testAudioChannelInAudioNodeStream();
+  is(channel, "normal", "AudioNodeStream is using the correct default audio channel.");
+
+  SpecialPowers.pushPermissions(
+    [{ "type": "audio-channel-" + aChannel, "allow": true, "context": document }],
+    function() {
+      var ac = new AudioContext(aChannel);
+      is(ac.mozAudioChannelType, aChannel, "Default ac channel == '" + aChannel + "'");
+
+      var channel = SpecialPowers.wrap(ac).testAudioChannelInAudioNodeStream();
+      is(channel, aChannel, "AudioNodeStream is using the correct new audio channel.");
+
+      runTest();
+    }
+  );
+}
+
+function test_preferences(aChannel) {
+  SpecialPowers.pushPrefEnv({"set": [["media.defaultAudioChannel", aChannel ]]},
+    function() {
+      SpecialPowers.pushPermissions(
+        [{ "type": "audio-channel-" + aChannel, "allow": false, "context": document }],
+        function() {
+          var ac = new AudioContext(aChannel);
+          ok(ac, "AudioContext created");
+          is(ac.mozAudioChannelType, aChannel, "Default ac channel == '" + aChannel + "'");
+
+          var channel = SpecialPowers.wrap(ac).testAudioChannelInAudioNodeStream();
+          is(channel, aChannel, "AudioNodeStream is using the correct audio channel.");
+
+          runTest();
+        }
+      );
+    }
+  );
+}
+
+function test_wrong_preferences() {
+  SpecialPowers.pushPrefEnv({"set": [["media.defaultAudioChannel", 'foobar' ]]},
+    function() {
+      var ac = new AudioContext();
+      ok(ac, "AudioContext created");
+      is(ac.mozAudioChannelType, 'normal', "Default ac channel == 'normal'");
+      runTest();
+    }
+  );
+}
+
+function test_testAudioChannelInAudioNodeStream() {
+  var ac = new AudioContext();
+  ok(ac, "AudioContext created");
+
+  var status = false;
+  try {
+    ac.testAudioChannelInAudioNodeStream();
+  } catch(e) {
+    status = true;
+  }
+
+  ok(status, "testAudioChannelInAudioNodeStream() should not exist in content.");
+  runTest();
+}
+
+var tests = [
+  test_basic,
+
+  function() { test_permission("content"); },
+  function() { test_permission("notification"); },
+  function() { test_permission("alarm"); },
+  function() { test_permission("telephony"); },
+  function() { test_permission("ringer"); },
+  function() { test_permission("publicnotification"); },
+
+  function() { test_preferences("content"); },
+  function() { test_preferences("notification"); },
+  function() { test_preferences("alarm"); },
+  function() { test_preferences("telephony"); },
+  function() { test_preferences("ringer"); },
+  function() { test_preferences("publicnotification"); },
+
+  test_wrong_preferences,
+
+  test_testAudioChannelInAudioNodeStream,
+];
+
+function runTest() {
+  if (!tests.length) {
+    SimpleTest.finish();
+    return;
+  }
+
+  var test = tests.shift();
+  test();
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.useAudioChannelAPI", true ]]}, runTest);
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestLongerTimeout(5);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_nodeToParamConnection.html b/dom/media/webaudio/test/test_nodeToParamConnection.html
new file mode 100644
index 0000000000..4525923db5
--- /dev/null
+++ b/dom/media/webaudio/test/test_nodeToParamConnection.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test connecting an AudioNode to an AudioParam</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  createGraph: function(context) {
+    var sourceBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+      sourceBuffer.getChannelData(1)[i] = -1;
+    }
+
+    var destination = context.destination;
+
+    var paramSource = context.createBufferSource();
+    paramSource.buffer = this.buffer;
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+
+    paramSource.connect(gain.gain);
+    source.connect(gain);
+
+    paramSource.start(0);
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      for (var j = 0; j < 2; ++j) {
+        this.buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+      }
+    }
+    var expectedBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 1 + (this.buffer.getChannelData(0)[i] + this.buffer.getChannelData(1)[i]) / 2;
+      expectedBuffer.getChannelData(1)[i] = -(1 + (this.buffer.getChannelData(0)[i] + this.buffer.getChannelData(1)[i]) / 2);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html b/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html
new file mode 100644
index 0000000000..6751066970
--- /dev/null
+++ b/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext with a channel count less than the specified number</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new OfflineAudioContext(2, 100, 22050);
+
+  var buf = ctx.createBuffer(6, 100, ctx.sampleRate);
+  for (var i = 0; i < 6; ++i) {
+    for (var j = 0; j < 100; ++j) {
+      buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+    }
+  }
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+  ctx.destination.channelCountMode = "max";
+  ctx.startRendering();
+  ctx.oncomplete = function(e) {
+    is(e.renderedBuffer.numberOfChannels, 2, "Correct expected number of buffers");
+    compareChannels(e.renderedBuffer.getChannelData(0), buf.getChannelData(0));
+    compareChannels(e.renderedBuffer.getChannelData(1), buf.getChannelData(1));
+
+    SimpleTest.finish();
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html b/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html
new file mode 100644
index 0000000000..7c7d5c8e52
--- /dev/null
+++ b/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext with a channel count less than the specified number</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new OfflineAudioContext(6, 100, 22050);
+
+  var buf = ctx.createBuffer(2, 100, ctx.sampleRate);
+  for (var i = 0; i < 2; ++i) {
+    for (var j = 0; j < 100; ++j) {
+      buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+    }
+  }
+  var emptyBuffer = ctx.createBuffer(1, 100, ctx.sampleRate);
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+  ctx.destination.channelCountMode = "max";
+  ctx.startRendering();
+  ctx.oncomplete = function(e) {
+    is(e.renderedBuffer.numberOfChannels, 6, "Correct expected number of buffers");
+    compareChannels(e.renderedBuffer.getChannelData(0), buf.getChannelData(0));
+    compareChannels(e.renderedBuffer.getChannelData(1), buf.getChannelData(1));
+    for (var i = 2; i < 6; ++i) {
+      compareChannels(e.renderedBuffer.getChannelData(i), emptyBuffer.getChannelData(0));
+    }
+
+    SimpleTest.finish();
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNode.html b/dom/media/webaudio/test/test_oscillatorNode.html
new file mode 100644
index 0000000000..5eb4885743
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNode.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+  var osc = context.createOscillator();
+
+  is(osc.channelCount, 2, "Oscillator node has 2 input channels by default");
+  is(osc.channelCountMode, "max", "Correct channelCountMode for the Oscillator node");
+  is(osc.channelInterpretation, "speakers", "Correct channelCountInterpretation for the Oscillator node");
+  is(osc.type, "sine", "Correct default type");
+  expectException(function() {
+    osc.type = "custom";
+  }, DOMException.INVALID_STATE_ERR);
+  is(osc.type, "sine", "Cannot set the type to custom");
+  is(osc.frequency.value, 440, "Correct default frequency value");
+  is(osc.detune.value, 0, "Correct default detine value");
+
+  // Make sure that we can set all of the valid type values
+  var types = [
+    "sine",
+    "square",
+    "sawtooth",
+    "triangle",
+  ];
+  for (var i = 0; i < types.length; ++i) {
+    osc.type = types[i];
+  }
+
+  // Verify setPeriodicWave()
+  var real = new Float32Array([1.0, 0.5, 0.25, 0.125]);
+  var imag = new Float32Array([1.0, 0.7, -1.0, 0.5]);
+  osc.setPeriodicWave(context.createPeriodicWave(real, imag));
+  is(osc.type, "custom", "Failed to set custom waveform");
+
+  expectNoException(function() {
+    osc.start();
+  });
+  expectNoException(function() {
+    osc.stop();
+  });
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNode2.html b/dom/media/webaudio/test/test_oscillatorNode2.html
new file mode 100644
index 0000000000..1ddae937c6
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNode2.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OscillatorNode lifetime and sine phase</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+
+function createOscillator(context) {
+  var osc = context.createOscillator();
+  osc.start(0);
+  osc.stop(signalLength/context.sampleRate);
+  return osc;
+}
+
+function connectUnreferencedOscillator(context, destination) {
+  var osc = createOscillator(context);
+  osc.connect(destination);
+}
+
+var gTest = {
+  length: signalLength,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var blend = context.createGain();
+
+    connectUnreferencedOscillator(context, blend);
+    // Test that the unreferenced oscillator remains alive until it has finished.
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+
+    // Create another sine wave oscillator in negative time, which should
+    // cancel when mixed with the unreferenced oscillator.
+    var oscillator = createOscillator(context);
+    oscillator.frequency.value = -440;
+    oscillator.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html b/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html
new file mode 100644
index 0000000000..8acc025da4
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode when the frequency is negative</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var types = ["sine",
+               "square",
+               "sawtooth",
+               "triangle"];
+
+  var finished = 0;
+  function finish() {
+    if (++finished == types.length) {
+      SimpleTest.finish();
+    }
+  }
+
+  types.forEach(function(t) {
+    var context = new OfflineAudioContext(1, 256, 44100);
+    var osc = context.createOscillator();
+
+    osc.frequency.value = -440;
+    osc.type = t;
+
+    osc.connect(context.destination);
+    osc.start();
+    context.startRendering().then(function(buffer) {
+      var samples = buffer.getChannelData(0);
+      // This samples the wave form in the middle of the first period, the value
+      // should be negative.
+      ok(samples[Math.floor(44100 / 440 / 4)] < 0., "Phase should be inverted when using a " + t + " waveform");
+      finish();
+    });
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodePassThrough.html b/dom/media/webaudio/test/test_oscillatorNodePassThrough.html
new file mode 100644
index 0000000000..c732bb2735
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodePassThrough.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test Oscillator with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createOscillator();
+
+    var srcWrapped = SpecialPowers.wrap(source);
+    ok("passThrough" in srcWrapped, "OscillatorNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [expectedBuffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodeStart.html b/dom/media/webaudio/test/test_oscillatorNodeStart.html
new file mode 100644
index 0000000000..c43219c993
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodeStart.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+  var osc = context.createOscillator();
+  var sp = context.createScriptProcessor(0, 1, 0);
+
+  osc.connect(sp);
+
+  sp.onaudioprocess = function (e) {
+    var input = e.inputBuffer.getChannelData(0);
+    var isSilent = true;
+    for (var i = 0; i < input.length; i++) {
+      if (input[i] != 0.0) {
+        isSilent = false;
+      }
+    }
+    sp.onaudioprocess = null;
+    ok(isSilent, "OscillatorNode should be silent before calling start.");
+    SimpleTest.finish();
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorTypeChange.html b/dom/media/webaudio/test/test_oscillatorTypeChange.html
new file mode 100644
index 0000000000..aaf311a0cb
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorTypeChange.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OscillatorNode type change after it has started and triangle phase</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const bufferSize = 1024;
+
+function startTest() {
+  var ctx = new AudioContext();
+
+  var oscillator1 = ctx.createOscillator();
+  oscillator1.connect(ctx.destination);
+  oscillator1.start(0);
+
+  // Assuming the above Web Audio operations have already scheduled an event
+  // to run in stable state and start the graph thread, schedule a subsequent
+  // event to change the type of oscillator1.
+  SimpleTest.executeSoon(function() {
+    oscillator1.type = "triangle";
+
+    // Another triangle wave with -1 gain should cancel the first.  This is
+    // starting at the same time as the type change, assuming that the phase
+    // is reset on type change.  A negative frequency should achieve the same
+    // as the -1 gain but for bug 916285.
+    var oscillator2 = ctx.createOscillator();
+    oscillator2.type = "triangle";
+    oscillator2.start(0);
+
+    var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+    oscillator1.connect(processor);
+    var gain = ctx.createGain();
+    gain.gain.value = -1;
+    gain.connect(processor);
+    oscillator2.connect(gain);
+
+    processor.onaudioprocess = function(e) {
+      compareChannels(e.inputBuffer.getChannelData(0),
+                      new Float32Array(bufferSize));
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+  });
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode.html b/dom/media/webaudio/test/test_pannerNode.html
new file mode 100644
index 0000000000..374ad34213
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode.html
@@ -0,0 +1,73 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-4, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var panner = context.createPanner();
+
+  source.buffer = buffer;
+
+  source.connect(panner);
+  panner.connect(destination);
+
+  // Verify default values
+  is(panner.panningModel, "equalpower", "Correct default value for panning model");
+  is(panner.distanceModel, "inverse", "Correct default value for distance model");
+  near(panner.refDistance, 1, "Correct default value for ref distance");
+  near(panner.maxDistance, 10000, "Correct default value for max distance");
+  near(panner.rolloffFactor, 1, "Correct default value for rolloff factor");
+  near(panner.coneInnerAngle, 360, "Correct default value for cone inner angle");
+  near(panner.coneOuterAngle, 360, "Correct default value for cone outer angle");
+  near(panner.coneOuterGain, 0, "Correct default value for cone outer gain");
+  is(panner.channelCount, 2, "panner node has 2 input channels by default");
+  is(panner.channelCountMode, "clamped-max", "Correct channelCountMode for the panner node");
+  is(panner.channelInterpretation, "speakers", "Correct channelCountInterpretation for the panner node");
+
+  panner.setPosition(1, 1, 1);
+  near(panner.positionX.value, 1, "setPosition sets AudioParam properly");
+  near(panner.positionY.value, 1, "setPosition sets AudioParam properly");
+  near(panner.positionZ.value, 1, "setPosition sets AudioParam properly");
+
+  panner.setOrientation(0, 1, 0);
+  near(panner.orientationX.value, 0, "setOrientation sets AudioParam properly");
+  near(panner.orientationY.value, 1, "setOrientation sets AudioParam properly");
+  near(panner.orientationZ.value, 0, "setOrientation sets AudioParam properly");
+
+  panner.setVelocity(1, 1, 1);
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+    panner.disconnect();
+
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeAbove.html b/dom/media/webaudio/test/test_pannerNodeAbove.html
new file mode 100644
index 0000000000..6bab394e66
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeAbove.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode directly above</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  numberOfChannels: 2,
+  createGraph: function(context) {
+    // An up vector will be made perpendicular to the front vector, in the
+    // front-up plane.
+    context.listener.setOrientation(0, 6.311749985202524e+307, 0, 0.1, 1000, 0);
+    // Linearly dependent vectors are ignored.
+    context.listener.setOrientation(0, 0, -6.311749985202524e+307, 0, 0, 6.311749985202524e+307);
+    var panner = context.createPanner();
+    panner.positionX.value = 2; // directly above
+    panner.rolloffFactor = 0; // no distance gain
+    panner.panningModel = "equalpower"; // no effect when directly above
+
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+    source.connect(panner);
+    source.start(0);
+
+    return panner;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      // Different signals in left and right buffers
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      expectedBuffer.getChannelData(1)[i] = Math.sin(220 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html b/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html
new file mode 100644
index 0000000000..21abd2b604
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode produces output even when the even when the distance is from the listener is zero</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var BUF_SIZE = 128;
+
+var types = [
+  "equalpower",
+  "HRTF"
+]
+
+var finished = types.length;
+
+function finish() {
+  if (!--finished) {
+    SimpleTest.finish();
+  }
+}
+
+function test(type) {
+  var ac = new OfflineAudioContext(1, BUF_SIZE, 44100);
+
+  // A sine to be used to fill the buffers
+  function sine(t) {
+    return Math.sin(440 * 2 * Math.PI * t / ac.sampleRate);
+  }
+
+  var monoBuffer = ac.createBuffer(1, BUF_SIZE, ac.sampleRate);
+  for (var i = 0; i < BUF_SIZE; ++i) {
+    monoBuffer.getChannelData(0)[i] = sine(i);
+  }
+
+  var monoSource = ac.createBufferSource();
+  monoSource.buffer = monoBuffer;
+  monoSource.start(0);
+
+  var panner = ac.createPanner();
+  panner.distanceModel = "linear";
+  panner.refDistance = 1;
+  panner.positionX.value = 0;
+  panner.positionY.value = 0;
+  panner.positionZ.value = 0;
+  monoSource.connect(panner);
+
+  var panner2 = ac.createPanner();
+  panner2.distanceModel = "inverse";
+  panner2.refDistance = 1;
+  panner2.positionX.value = 0;
+  panner2.positionY.value = 0;
+  panner2.positionZ.value = 0;
+  panner.connect(panner2);
+
+  var panner3 = ac.createPanner();
+  panner3.distanceModel = "exponential";
+  panner3.refDistance = 1;
+  panner3.positionX.value = 0;
+  panner3.positionY.value = 0;
+  panner3.positionZ.value = 0;
+  panner2.connect(panner3);
+
+  panner3.connect(ac.destination);
+
+  ac.startRendering().then(function(buffer) {
+    compareBuffers(buffer, monoBuffer);
+    finish();
+  });
+}
+
+addLoadEvent(function() {
+  types.forEach(test);
+});
+
+SimpleTest.waitForExplicitFinish();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeChannelCount.html b/dom/media/webaudio/test/test_pannerNodeChannelCount.html
new file mode 100644
index 0000000000..63a52ea0cc
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeChannelCount.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode directly above</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 2,
+  createGraph: function(context) {
+    var buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var sample = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      // When mixed into a single channel, this produces silence
+      buffer.getChannelData(0)[i] = sample;
+      buffer.getChannelData(1)[i] = -sample;
+    }
+
+    var panner = context.createPanner();
+    panner.positionX.value = 1;
+    panner.positionY.value = 2;
+    panner.positionZ.value = 3;
+    panner.channelCount = 1;
+    expectException(function() { panner.channelCount = 3; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    panner.channelCountMode = "explicit";
+    expectException(function() { panner.channelCountMode = "max"; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    panner.channelInterpretation = "discrete";
+    panner.channelInterpretation = "speakers";
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(panner);
+    source.start(0);
+
+    return panner;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html b/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html
new file mode 100644
index 0000000000..c5312d0421
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html
@@ -0,0 +1,106 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test left/right symmetry and block-offset invariance of HRTF panner</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const blockSize = 128;
+const bufferSize = 4096; // > HRTF panner latency
+
+var ctx = new AudioContext();
+
+function isChannelSilent(channel) {
+  for (var i = 0; i < channel.length; ++i) {
+    if (channel[i] != 0.0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+function startTest() {
+  var leftPanner = ctx.createPanner();
+  var rightPanner = ctx.createPanner();
+  leftPanner.type = "HRTF";
+  rightPanner.type = "HRTF";
+  leftPanner.positionX.value = -1;
+  rightPanner.positionX.value = 1;
+
+  // Test that PannerNode processes the signal consistently irrespective of
+  // the offset in the processing block.  This is done by inserting a delay of
+  // less than a block size before one panner.
+  const delayTime = 0.7 * blockSize / ctx.sampleRate;
+  var leftDelay = ctx.createDelay(delayTime);
+  leftDelay.delayTime.value = delayTime;
+  leftDelay.connect(leftPanner);
+  // and compensating for the delay after the other.
+  var rightDelay = ctx.createDelay(delayTime);
+  rightDelay.delayTime.value = delayTime;
+  rightPanner.connect(rightDelay);
+
+  // Feed the panners with a signal having some harmonics to fill the spectrum.
+  var oscillator = ctx.createOscillator();
+  oscillator.frequency.value = 110;
+  oscillator.type = "sawtooth";
+  oscillator.connect(leftDelay);
+  oscillator.connect(rightPanner);
+  oscillator.start(0);
+
+  // Switch the channels on one panner output, and it should match the other.
+  var splitter = ctx.createChannelSplitter();
+  leftPanner.connect(splitter);
+  var merger = ctx.createChannelMerger();
+  splitter.connect(merger, 0, 1);
+  splitter.connect(merger, 1, 0);
+
+  // Invert one signal so that mixing with the other will find the difference.
+  var gain = ctx.createGain();
+  gain.gain.value = -1.0;
+  merger.connect(gain);
+
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  gain.connect(processor);
+  rightDelay.connect(processor);
+  processor.onaudioprocess =
+    function(e) {
+      compareBuffers(e.inputBuffer,
+                     ctx.createBuffer(2, bufferSize, ctx.sampleRate));
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+}
+
+function prepareTest() {
+  // A PannerNode will produce no output until it has loaded its HRIR
+  // database.  Wait for this to load before starting the test.
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  var panner = ctx.createPanner();
+  panner.connect(processor);
+  var oscillator = ctx.createOscillator();
+  oscillator.connect(panner);
+  oscillator.start(0);
+
+  processor.onaudioprocess =
+    function(e) {
+      if (isChannelSilent(e.inputBuffer.getChannelData(0)))
+        return;
+
+      oscillator.stop(0);
+      panner.disconnect();
+      e.target.onaudioprocess = null;
+      startTest();
+    };
+}
+prepareTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodePassThrough.html b/dom/media/webaudio/test/test_pannerNodePassThrough.html
new file mode 100644
index 0000000000..ab1f4b46fb
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodePassThrough.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var panner = context.createPanner();
+
+    source.buffer = this.buffer;
+
+    source.connect(panner);
+
+    context.listener.setOrientation(0, 6.311749985202524e+307, 0, 0.1, 1000, 0);
+    context.listener.setOrientation(0, 0, -6.311749985202524e+307, 0, 0, 6.311749985202524e+307);
+    panner.positionX = 2;
+    panner.rolloffFactor = 0;
+    panner.panningModel = "equalpower";
+
+    var pannerWrapped = SpecialPowers.wrap(panner);
+    ok("passThrough" in pannerWrapped, "PannerNode should support the passThrough API");
+    pannerWrapped.passThrough = true;
+
+    source.start(0);
+    return panner;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeTail.html b/dom/media/webaudio/test/test_pannerNodeTail.html
new file mode 100644
index 0000000000..5fff527972
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeTail.html
@@ -0,0 +1,232 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of PannerNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// This tests that a PannerNode does not release its reference before
+// it finishes emitting sound.
+//
+// The PannerNode tail time is short, so, when a PannerNode is destroyed on
+// the main thread, it is unlikely to notify the graph thread before the tail
+// time expires.  However, by adding DelayNodes downstream from the
+// PannerNodes, the graph thread can have enough time to notice that a
+// DelayNode has been destroyed.
+//
+// In the current implementation, DelayNodes will take a tail-time reference
+// immediately when they receive the first block of sound from an upstream
+// node, so this test connects the downstream DelayNodes while the upstream
+// nodes are finishing, and then runs GC (on the main thread) before the
+// DelayNodes receive any input (on the graph thread).
+//
+// Web Audio doesn't provide a means to precisely time connect()s but we can
+// test that the output of delay nodes matches the output from a reference
+// PannerNode that we know will not be GCed.
+//
+// Another set of delay nodes is added upstream to ensure that the source node
+// has removed its self-reference after dispatching its "ended" event.
+
+SimpleTest.waitForExplicitFinish();
+
+const blockSize = 128;
+// bufferSize should be long enough that to allow an audioprocess event to be
+// sent to the main thread and a connect message to return to the graph
+// thread.
+const bufferSize = 4096;
+const pannerCount = bufferSize / blockSize;
+// sourceDelayBufferCount should be long enough to allow the source node
+// onended to finish and remove the source self-reference.
+const sourceDelayBufferCount = 3;
+var gotEnded = false;
+// ccDelayLength should be long enough to allow CC to run
+var ccDelayBufferCount = 20;
+const ccDelayLength = ccDelayBufferCount * bufferSize;
+
+var ctx;
+var testPanners = [];
+var referencePanner;
+var referenceProcessCount = 0;
+var referenceOutput = [new Float32Array(bufferSize),
+                       new Float32Array(bufferSize)];
+var testProcessor;
+var testProcessCount = 0;
+
+function isChannelSilent(channel) {
+  for (var i = 0; i < channel.length; ++i) {
+    if (channel[i] != 0.0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+function onReferenceOutput(e) {
+  switch(referenceProcessCount) {
+
+  case sourceDelayBufferCount - 1:
+    // The panners are about to finish.
+    if (!gotEnded) {
+      todo(false, "Source hasn't ended.  Increase sourceDelayBufferCount?");
+    }
+
+    // Connect each PannerNode output to a downstream DelayNode,
+    // and connect ScriptProcessors to compare test and reference panners.
+    var delayDuration = ccDelayLength / ctx.sampleRate;
+    for (var i = 0; i < pannerCount; ++i) {
+      var delay = ctx.createDelay(delayDuration);
+      delay.delayTime.value = delayDuration;
+      delay.connect(testProcessor);
+      testPanners[i].connect(delay);
+    }
+    testProcessor = null;
+    testPanners = null;
+
+    // The panning effect is linear so only one reference panner is required.
+    // This also checks that the individual panners don't chop their output
+    // too soon.
+    referencePanner.connect(e.target);
+
+    // Assuming the above operations have already scheduled an event to run in
+    // stable state and ask the graph thread to make connections, schedule a
+    // subsequent event to run cycle collection, which should not collect
+    // panners that are still producing sound.
+    SimpleTest.executeSoon(function() {
+      SpecialPowers.forceGC();
+      SpecialPowers.forceCC();
+    });
+
+    break;
+
+  case sourceDelayBufferCount:
+    // Record this buffer during which PannerNode outputs were connected.
+    for (var i = 0; i < 2; ++i) {
+      e.inputBuffer.copyFromChannel(referenceOutput[i], i);
+    }
+    e.target.onaudioprocess = null;
+    e.target.disconnect();
+
+    // If the buffer is silent, there is probably not much point just
+    // increasing the buffer size, because, with the buffer size already
+    // significantly larger than panner tail time, it demonstrates that the
+    // lag between threads is much greater than the tail time.
+    if (isChannelSilent(referenceOutput[0])) {
+      todo(false, "Connections not detected.");
+    }
+  }
+
+  referenceProcessCount++;
+}
+
+function onTestOutput(e) {
+  if (testProcessCount < sourceDelayBufferCount + ccDelayBufferCount) {
+    testProcessCount++;
+    return;
+  }
+
+  for (var i = 0; i < 2; ++i) {
+    compareChannels(e.inputBuffer.getChannelData(i), referenceOutput[i]);
+  }
+  e.target.onaudioprocess = null;
+  e.target.disconnect();
+  SimpleTest.finish();
+}
+
+function startTest() {
+  // 0.002 is MaxDelayTimeSeconds in HRTFpanner.cpp
+  // and 512 is fftSize() at 48 kHz.
+  const expectedPannerTailTime = 0.002 * ctx.sampleRate + 512;
+
+  // Create some PannerNodes downstream from DelayNodes with delays long
+  // enough for their source to finish, dispatch its "ended" event
+  // and release its playing reference.  The DelayNodes should expire their
+  // tail-time references before the PannerNodes and so only the PannerNode
+  // lifetimes depends on their tail-time references.  Many DelayNodes are
+  // created and timed to finish at different times so that one PannerNode
+  // will be finishing the block processed immediately after the connect is
+  // received.
+  var source = ctx.createBufferSource();
+  // Just short of blockSize here to avoid rounding into the next block
+  var buffer = ctx.createBuffer(1, blockSize - 1, ctx.sampleRate);
+  for (var i = 0; i < buffer.length; ++i) {
+    buffer.getChannelData(0)[i] = Math.cos(Math.PI * i / buffer.length);
+  }
+  source.buffer = buffer;
+  source.start(0);
+  source.onended = function(e) {
+    gotEnded = true;
+  };
+
+  // Time the first test panner to finish just before downstream DelayNodes
+  // are about the be connected.  Note that DelayNode lifetime depends on
+  // maxDelayTime so set that equal to the delay.
+  var delayDuration =
+    (sourceDelayBufferCount * bufferSize
+     - expectedPannerTailTime - 2 * blockSize) / ctx.sampleRate;
+
+  for (var i = 0; i < pannerCount; ++i) {
+    var delay = ctx.createDelay(delayDuration);
+    delay.delayTime.value = delayDuration;
+    source.connect(delay);
+    delay.connect(referencePanner)
+
+    var panner = ctx.createPanner();
+    panner.type = "HRTF";
+    delay.connect(panner);
+    testPanners[i] = panner;
+
+    delayDuration += blockSize / ctx.sampleRate;
+  }
+
+  // Create a ScriptProcessor now to use as a timer to trigger connection of
+  // downstream nodes.  It will also be used to record reference output.
+  var referenceProcessor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  referenceProcessor.onaudioprocess = onReferenceOutput;
+  // Start audioprocess events before source delays are connected.
+  referenceProcessor.connect(ctx.destination);
+
+  // The test ScriptProcessor will record output of testPanners. 
+  // Create it now so that it is synchronized with the referenceProcessor.
+  testProcessor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  testProcessor.onaudioprocess = onTestOutput;
+  // Start audioprocess events before source delays are connected.
+  testProcessor.connect(ctx.destination);
+}
+
+function prepareTest() {
+  ctx = new AudioContext();
+  // Place the listener to the side of the origin, where the panners are
+  // positioned, to maximize delay in one ear.
+  ctx.listener.setPosition(1,0,0);
+
+  // A PannerNode will produce no output until it has loaded its HRIR
+  // database.  Wait for this to load before starting the test.
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  referencePanner = ctx.createPanner();
+  referencePanner.type = "HRTF";
+  referencePanner.connect(processor);
+  var oscillator = ctx.createOscillator();
+  oscillator.connect(referencePanner);
+  oscillator.start(0);
+
+  processor.onaudioprocess = function(e) {
+    if (isChannelSilent(e.inputBuffer.getChannelData(0)))
+      return;
+
+    oscillator.stop(0);
+    oscillator.disconnect();
+    referencePanner.disconnect();
+    e.target.onaudioprocess = null;
+    SimpleTest.executeSoon(startTest);
+  };
+}
+prepareTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode_equalPower.html b/dom/media/webaudio/test/test_pannerNode_equalPower.html
new file mode 100644
index 0000000000..14e9f2153c
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode_equalPower.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<title>Test PannerNode</title>
+<script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+<script type="text/javascript" src="webaudio.js"></script>
+<script type="text/javascript" src="layouttest-glue.js"></script>
+<script type="text/javascript" src="blink/audio-testing.js"></script>
+<script type="text/javascript" src="blink/panner-model-testing.js"></script>
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  function checkFinished() {
+    SimpleTest.finish();
+  }
+  var ctx = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+  createTestAndRun(ctx, nodesToCreate, 2, checkFinished);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode_maxDistance.html b/dom/media/webaudio/test/test_pannerNode_maxDistance.html
new file mode 100644
index 0000000000..faca136b3a
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode_maxDistance.html
@@ -0,0 +1,64 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode outputs silence when the distance is greater than maxDist</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var types = [
+  "equalpower",
+  "HRTF"
+]
+
+var finished = types.length;
+
+function finish() {
+  if (!--finished) {
+    SimpleTest.finish();
+  }
+}
+
+function test(type) {
+  var ac = new OfflineAudioContext(1, 128, 44100);
+  var osc = ac.createOscillator();
+  var panner = ac.createPanner();
+
+  panner.distanceModel = "linear";
+  panner.maxDistance = 100;
+  panner.positionY.value = 200;
+  ac.listener.setPosition(0, 0, 0);
+
+  osc.connect(panner);
+  panner.connect(ac.destination);
+
+  osc.start();
+
+  ac.startRendering().then(function(buffer) {
+    var silence = true;
+    var array = buffer.getChannelData(0);
+    for (var i = 0; i < buffer.length; i++) {
+      if (array[i] != 0) {
+        ok(false, "Found noise in the buffer.");
+        silence = false;
+      }
+    }
+    ok(silence, "The buffer is silent.");
+    finish();
+  });
+}
+
+
+addLoadEvent(function() {
+  types.forEach(test);
+});
+
+SimpleTest.waitForExplicitFinish();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_periodicWave.html b/dom/media/webaudio/test/test_periodicWave.html
new file mode 100644
index 0000000000..3ed4407486
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWave.html
@@ -0,0 +1,94 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the PeriodicWave interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// real and imag are used in separate PeriodicWaves to make their peak values
+// easy to determine.
+const realMax = 99;
+var real = new Float32Array(realMax + 1);
+real[1] = 2.0; // fundamental
+real[realMax] = 3.0;
+const realPeak = real[1] + real[realMax];
+const realFundamental = 19.0;
+var imag = new Float32Array(4);
+imag[0] = 6.0; // should be ignored.
+imag[3] = 0.5;
+const imagPeak = imag[3];
+const imagFundamental = 551.0;
+
+const testLength = 4096;
+
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ac.createPeriodicWave(new Float32Array(4096), new Float32Array(4096));
+  expectException(function() {
+    ac.createPeriodicWave(new Float32Array(512), imag);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    ac.createPeriodicWave(new Float32Array(0), new Float32Array(0));
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectNoException(function() {
+    ac.createPeriodicWave(new Float32Array(4097), new Float32Array(4097));
+  });
+
+  runTest();
+});
+
+var gTest = {
+  createGraph: function(context) {
+    var merger = context.createChannelMerger();
+
+    var osc0 = context.createOscillator();
+    var osc1 = context.createOscillator();
+
+    osc0.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length)));
+    osc1.setPeriodicWave(context.
+                         createPeriodicWave(new Float32Array(imag.length),
+                                            imag));
+
+    osc0.frequency.value = realFundamental;
+    osc1.frequency.value = imagFundamental;
+
+    osc0.start();
+    osc1.start();
+
+    osc0.connect(merger, 0, 0);
+    osc1.connect(merger, 0, 1);
+
+    return merger;
+  },
+  createExpectedBuffers: function(context) {
+    var buffer = context.createBuffer(2, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+
+      buffer.getChannelData(0)[i] = 1.0 / realPeak *
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+
+      buffer.getChannelData(1)[i] = 1.0 / imagPeak *
+         imag[3] * Math.sin(2 * Math.PI * 3 * imagFundamental * i /
+                            context.sampleRate);
+    }
+    return buffer;
+  },
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_periodicWaveBandLimiting.html b/dom/media/webaudio/test/test_periodicWaveBandLimiting.html
new file mode 100644
index 0000000000..70fbb09e2a
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWaveBandLimiting.html
@@ -0,0 +1,86 @@
+<!DOCTYPE html>
+<title>Test effect of band limiting on PeriodicWave signals</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+const sampleRate = 48000;
+const bufferSize = 12800;
+const epsilon = 0.01;
+
+// "All implementations must support arrays up to at least 8192", but the
+// linear interpolation of the current implementation distorts the higher
+// frequency components too much to pass this test.
+const frequencyIndexMax = 200;
+
+// A set of oscillators are created near the Nyquist frequency.
+// These are factors giving each oscillator frequency relative to the Nyquist.
+// The first is an octave below Nyquist and the last is just above.
+const OCTAVE_BELOW = 0;
+const HALF_BELOW = 1;
+const NEAR_BELOW = 2;
+const ABOVE = 3;
+const oscillatorFactors = [0.5, Math.sqrt(0.5), 0.99, 1.01];
+const oscillatorCount = oscillatorFactors.length;
+
+// Return magnitude relative to unit sine wave
+function magnitude(array) {
+  var mag = 0
+  for (var i = 0; i < array.length; ++i) {
+    sample = array[i];
+    mag += sample * sample;
+  }
+  return Math.sqrt(2 * mag / array.length);
+}
+
+function test_frequency_index(frequencyIndex) {
+
+  var context =
+    new OfflineAudioContext(oscillatorCount, bufferSize, sampleRate);
+
+  var merger = context.createChannelMerger(oscillatorCount);
+  merger.connect(context.destination);
+
+  var real = new Float32Array(frequencyIndex + 1);
+  real[frequencyIndex] = 1;
+  var image = new Float32Array(real.length);
+  var wave = context.createPeriodicWave(real, image);
+
+  for (var i = 0; i < oscillatorCount; ++i) {
+    var oscillator = context.createOscillator();
+    oscillator.frequency.value =
+      oscillatorFactors[i] * sampleRate / (2 * frequencyIndex);
+    oscillator.connect(merger, 0, i);
+    oscillator.setPeriodicWave(wave);
+    oscillator.start(0);
+  }
+
+  return context.startRendering().
+    then((buffer) => {
+      assert_equals(buffer.numberOfChannels, oscillatorCount);
+      var magnitudes = [];
+      for (var i = 0; i < oscillatorCount; ++i) {
+        magnitudes[i] = magnitude(buffer.getChannelData(i));
+      }
+      // Unaffected by band-limiting one octave below Nyquist.
+      assert_approx_equals(magnitudes[OCTAVE_BELOW], 1, epsilon,
+                           "magnitude with frequency octave below Nyquist");
+      // Still at least half the amplitude at half octave below Nyquist.
+      assert_greater_than(magnitudes[HALF_BELOW], 0.5 * (1 - epsilon),
+                          "magnitude with frequency half octave below Nyquist");
+      // Approaching zero or zero near Nyquist.
+      assert_less_than(magnitudes[NEAR_BELOW], 0.1,
+                       "magnitude with frequency near Nyquist");
+      assert_equals(magnitudes[ABOVE], 0,
+                   "magnitude with frequency above Nyquist");
+    });
+}
+
+// The 5/4 ratio with rounding up provides sampling across a range of
+// octaves and offsets within octaves.
+for (var frequencyIndex = 1;
+     frequencyIndex < frequencyIndexMax;
+     frequencyIndex = Math.floor((5 * frequencyIndex + 3) / 4)) {
+  promise_test(test_frequency_index.bind(null, frequencyIndex),
+               "Frequency " + frequencyIndex);
+}
+</script>
diff --git a/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html b/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html
new file mode 100644
index 0000000000..fb924c4751
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html
@@ -0,0 +1,100 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PeriodicWave disableNormalization Parameter</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// We create PerodicWave instances containing two tones and compare it to
+// buffers created directly in JavaScript by adding the two waves together.
+// Two of the PeriodicWaves are normalized, the other is not. This test is
+// a modification of test_periodicWave.html.
+//
+// These constants are borrowed from test_periodicWave.html and modified
+// so that the realPeak (which is the normalization factor) will be small
+// enough that the errors are within the bounds for the test.
+const realMax = 99;
+var real = new Float32Array(realMax + 1);
+real[1] = 2.0; // fundamental
+real[realMax] = 0.25;
+
+const realPeak = real[1] + real[realMax];
+const realFundamental = 19.0;
+
+const testLength = 4096;
+
+addLoadEvent(function() {
+  runTest();
+});
+
+var gTest = {
+  createGraph: function(context) {
+    var merger = context.createChannelMerger();
+
+    var osc0 = context.createOscillator();
+    var osc1 = context.createOscillator();
+    var osc2 = context.createOscillator();
+
+    osc0.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length),
+                                            {disableNormalization: false}));
+    osc1.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length)));
+    osc2.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length),
+                                            {disableNormalization: true}));
+
+    osc0.frequency.value = realFundamental;
+    osc1.frequency.value = realFundamental;
+    osc2.frequency.value = realFundamental;
+
+    osc0.start();
+    osc1.start();
+    osc2.start();
+
+    osc0.connect(merger, 0, 0);
+    osc1.connect(merger, 0, 1);
+    osc2.connect(merger, 0, 2);
+
+    return merger;
+  },
+  createExpectedBuffers: function(context) {
+    var buffer = context.createBuffer(3, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+
+      buffer.getChannelData(0)[i] = 1.0 / realPeak *
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+
+      buffer.getChannelData(1)[i] = buffer.getChannelData(0)[i];
+
+      // TODO: We need to scale by a factor of two to make the results work
+      //       out here. This seems suspicious, see Bug 1266737.
+      buffer.getChannelData(2)[i] = 2.0 *
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+    }
+    return buffer;
+  },
+  'numberOfChannels': 3,
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNode.html b/dom/media/webaudio/test/test_scriptProcessorNode.html
new file mode 100644
index 0000000000..7cfb3d96e5
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNode.html
@@ -0,0 +1,132 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = null;
+
+  var sourceSP = context.createScriptProcessor(2048);
+  sourceSP.addEventListener("audioprocess", function(e) {
+    // generate the audio
+    for (var i = 0; i < 2048; ++i) {
+      // Make sure our first sample won't be zero
+      e.outputBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i + 1) / context.sampleRate);
+      e.outputBuffer.getChannelData(1)[i] = Math.sin(880 * 2 * Math.PI * (i + 1) / context.sampleRate);
+    }
+    // Remember our generated audio
+    buffer = e.outputBuffer;
+
+    sourceSP.removeEventListener("audioprocess", arguments.callee);
+  }, false);
+
+  expectException(function() {
+    context.createScriptProcessor(1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(2);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(128);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(255);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  is(sourceSP.channelCount, 2, "script processor node has 2 input channels by default");
+  is(sourceSP.channelCountMode, "explicit", "Correct channelCountMode for the script processor node");
+  is(sourceSP.channelInterpretation, "speakers", "Correct channelCountInterpretation for the script processor node");
+
+  function findFirstNonZeroSample(buffer) {
+    for (var i = 0; i < buffer.length; ++i) {
+      if (buffer.getChannelData(0)[i] != 0) {
+        return i;
+      }
+    }
+    return buffer.length;
+  }
+
+  var sp = context.createScriptProcessor(2048);
+  sourceSP.connect(sp);
+  sp.connect(context.destination);
+  var lastPlaybackTime = 0;
+
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  function checkAudioProcessingEvent(e) {
+    is(e.target, sp, "Correct event target");
+    ok(e.playbackTime > lastPlaybackTime, "playbackTime correctly set");
+    lastPlaybackTime = e.playbackTime;
+    is(e.inputBuffer.numberOfChannels, 2, "Correct number of channels for the input buffer");
+    is(e.inputBuffer.length, 2048, "Correct length for the input buffer");
+    is(e.inputBuffer.sampleRate, context.sampleRate, "Correct sample rate for the input buffer");
+    is(e.outputBuffer.numberOfChannels, 2, "Correct number of channels for the output buffer");
+    is(e.outputBuffer.length, 2048, "Correct length for the output buffer");
+    is(e.outputBuffer.sampleRate, context.sampleRate, "Correct sample rate for the output buffer");
+
+    compareChannels(e.outputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.outputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+  }
+
+  sp.onaudioprocess = function(e) {
+    isnot(buffer, null, "The audioprocess handler for sourceSP must be run at this point");
+    checkAudioProcessingEvent(e);
+
+    // Because of the initial latency added by the second script processor node,
+    // we will never see any generated audio frames in the first callback.
+    compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+
+    sp.onaudioprocess = function(e) {
+      checkAudioProcessingEvent(e);
+
+      var firstNonZero = findFirstNonZeroSample(e.inputBuffer);
+      ok(firstNonZero <= 2048, "First non-zero sample within range");
+
+      compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), 2048 - firstNonZero, firstNonZero, 0);
+      compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), 2048 - firstNonZero, firstNonZero, 0);
+
+      if (firstNonZero == 0) {
+        // If we did not experience any delays, the test is done!
+        sp.onaudioprocess = null;
+
+        SimpleTest.finish();
+      } else if (firstNonZero != 2048) {
+        // In case we just saw a zero buffer this time, wait one more round
+        sp.onaudioprocess = function(e) {
+          checkAudioProcessingEvent(e);
+
+          compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+
+          sp.onaudioprocess = null;
+
+          SimpleTest.finish();
+        };
+      }
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html b/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html
new file mode 100644
index 0000000000..6361a1747b
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html
@@ -0,0 +1,80 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(6, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    for (var j = 0; j < 6; ++j) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * j * Math.PI * i / context.sampleRate);
+    }
+  }
+
+  var monoBuffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    monoBuffer.getChannelData(0)[i] = 1;
+  }
+
+  var source = context.createBufferSource();
+
+  var sp = context.createScriptProcessor(2048, 3);
+  expectException(function() { sp.channelCount = 2; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  sp.channelCountMode = "explicit";
+  expectException(function() { sp.channelCountMode = "max"; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() { sp.channelCountMode = "clamped-max"; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  sp.channelInterpretation = "discrete";
+  source.start(0);
+  source.buffer = buffer;
+  source.connect(sp);
+  sp.connect(context.destination);
+
+  var monoSource = context.createBufferSource();
+  monoSource.buffer = monoBuffer;
+  monoSource.connect(sp);
+  monoSource.start(2048 / context.sampleRate);
+
+  sp.onaudioprocess = function(e) {
+    is(e.inputBuffer.numberOfChannels, 3, "Should be correctly down-mixed to three channels");
+    for (var i = 0; i < 3; ++i) {
+      compareChannels(e.inputBuffer.getChannelData(i), buffer.getChannelData(i));
+    }
+
+    // On the next iteration, we'll get a silence buffer
+    sp.onaudioprocess = function(e) {
+      var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+      is(e.inputBuffer.numberOfChannels, 3, "Should be correctly up-mixed to three channels");
+      compareChannels(e.inputBuffer.getChannelData(0), monoBuffer.getChannelData(0));
+      for (var i = 1; i < 3; ++i) {
+        compareChannels(e.inputBuffer.getChannelData(i), emptyBuffer.getChannelData(0));
+      }
+
+      sp.onaudioprocess = null;
+      sp.disconnect(context.destination);
+
+      SimpleTest.finish();
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html b/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html
new file mode 100644
index 0000000000..a3c073e382
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode: should not fire audioprocess if not connected.</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while to ensure that a given event does not happen.");
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var sp = context.createScriptProcessor(2048, 2, 2);
+  sp.onaudioprocess = function(e) {
+    ok(false, "Should not call onaudioprocess if the node is not connected.");
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  };
+  setTimeout(function() {
+    console.log(sp.onaudioprocess);
+    if (sp.onaudioprocess) {
+      ok(true, "onaudioprocess not fired.");
+      SimpleTest.finish();
+    }
+  }, 4000);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html b/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html
new file mode 100644
index 0000000000..8352a331da
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html
@@ -0,0 +1,103 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = null;
+
+  var sourceSP = context.createScriptProcessor(2048);
+  sourceSP.addEventListener("audioprocess", function(e) {
+    // generate the audio
+    for (var i = 0; i < 2048; ++i) {
+      // Make sure our first sample won't be zero
+      e.outputBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i + 1) / context.sampleRate);
+      e.outputBuffer.getChannelData(1)[i] = Math.sin(880 * 2 * Math.PI * (i + 1) / context.sampleRate);
+    }
+    // Remember our generated audio
+    buffer = e.outputBuffer;
+
+    sourceSP.removeEventListener("audioprocess", arguments.callee);
+  }, false);
+
+  function findFirstNonZeroSample(buffer) {
+    for (var i = 0; i < buffer.length; ++i) {
+      if (buffer.getChannelData(0)[i] != 0) {
+        return i;
+      }
+    }
+    return buffer.length;
+  }
+
+  var sp = context.createScriptProcessor(2048);
+  sourceSP.connect(sp);
+
+  var spWrapped = SpecialPowers.wrap(sp);
+  ok("passThrough" in spWrapped, "ScriptProcessorNode should support the passThrough API");
+  spWrapped.passThrough = true;
+
+  sp.onaudioprocess = function() {
+    ok(false, "The audioprocess event must never be dispatched on the passthrough ScriptProcessorNode");
+  };
+
+  var sp2 = context.createScriptProcessor(2048);
+  sp.connect(sp2);
+  sp2.connect(context.destination);
+
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  sp2.onaudioprocess = function(e) {
+    // Because of the initial latency added by the second script processor node,
+    // we will never see any generated audio frames in the first callback.
+    compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+
+    sp2.onaudioprocess = function(e) {
+      var firstNonZero = findFirstNonZeroSample(e.inputBuffer);
+      ok(firstNonZero <= 2048, "First non-zero sample within range");
+
+      compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), 2048 - firstNonZero, firstNonZero, 0);
+      compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), 2048 - firstNonZero, firstNonZero, 0);
+
+      if (firstNonZero == 0) {
+        // If we did not experience any delays, the test is done!
+        sp2.onaudioprocess = null;
+
+        SimpleTest.finish();
+      } else if (firstNonZero != 2048) {
+        // In case we just saw a zero buffer this time, wait one more round
+        sp2.onaudioprocess = function(e) {
+          compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+
+          sp2.onaudioprocess = null;
+
+          SimpleTest.finish();
+        };
+      }
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html b/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html
new file mode 100644
index 0000000000..6ac8beda0a
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var sp = context.createScriptProcessor(2048, 0, 2);
+  sp.onaudioprocess = function(e) {
+    is(e.inputBuffer.numberOfChannels, 0, "Should have 0 input channels");
+    is(e.outputBuffer.numberOfChannels, 2, "Should have 2 output channels");
+    sp.onaudioprocess = null;
+
+    sp = context.createScriptProcessor(2048, 2, 0);
+    sp.onaudioprocess = function(e) {
+      is(e.inputBuffer.numberOfChannels, 2, "Should have 2 input channels");
+      is(e.outputBuffer.numberOfChannels, 0, "Should have 0 output channels");
+      sp.onaudioprocess = null;
+
+      SimpleTest.finish();
+    };
+    sp.connect(context.destination);
+  };
+  sp.connect(context.destination);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html b/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html
new file mode 100644
index 0000000000..43cd139123
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode playbackTime for bug 970773</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var context = new AudioContext();
+const delay = 0.1;
+
+function doTest() {
+  const processorBufferLength = 256;
+  // |currentTime| may include double precision floating point
+  // rounding errors, so round to nearest integer sample to ignore these.
+  var minimumPlaybackSample =
+    Math.round(context.currentTime * context.sampleRate) +
+    processorBufferLength;
+  var sp = context.createScriptProcessor(processorBufferLength);
+  sp.connect(context.destination);
+  sp.onaudioprocess =
+    function(e) {
+      is(e.inputBuffer.length, processorBufferLength,
+         "expected buffer length");
+      var playbackSample = Math.round(e.playbackTime * context.sampleRate)
+      ok(playbackSample >= minimumPlaybackSample,
+         "playbackSample " + playbackSample +
+         " beyond expected minimum " + minimumPlaybackSample);
+      sp.onaudioprocess = null;
+      SimpleTest.finish();
+    };
+}
+
+// Wait until AudioDestinationNode has accumulated enough 'extra' time so that
+// a failure would be easily detected.
+(function waitForExtraTime() {
+  if (context.currentTime < delay) {
+    SimpleTest.executeSoon(waitForExtraTime);
+  } else {
+    doTest();
+  }
+})();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html
new file mode 100644
index 0000000000..5c03a8a911
--- /dev/null
+++ b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html
@@ -0,0 +1,72 @@
+<!DOCTYPE html>
+<title>Test seamless playback of a series of resampled buffers</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+// Permitting some accumulation of rounding to int16_t.
+// 64/2^15 would be only just small enough to detect off-by-one-subsample
+// scheduling errors with the frequencies here.
+const EPSILON = 4.0 / Math.pow(2, 15);
+// Offsets test for rounding to nearest rather than up or down.
+const OFFSETS = [EPSILON, 1.0 - EPSILON];
+// The ratio of resampling is 147:160, so 256 start points is enough to cover
+// every fractional offset.
+const LENGTH = 256;
+
+function do_test(context_rate, buffer_rate, start_offset) {
+
+  var context =
+    new OfflineAudioContext(2, LENGTH, context_rate);
+
+  var merger = context.createChannelMerger(context.destination.channelCount);
+  merger.connect(context.destination);
+
+  // Create an audio signal that will be repeated
+  var repeating_signal = context.createBuffer(1, 1, buffer_rate);
+  repeating_signal.getChannelData(0)[0] = 0.5;
+
+  // Schedule a series of nodes to repeat the signal.
+  for (var i = 0; i < LENGTH; ++i) {
+    var source = context.createBufferSource();
+    source.buffer = repeating_signal;
+    source.connect(merger, 0, 0);
+    source.start((i + start_offset) / buffer_rate);
+  }
+
+  // A single long signal should produce the same result.
+  var long_signal = context.createBuffer(1, LENGTH, buffer_rate);
+  var c = long_signal.getChannelData(0);
+  for (var i = 0; i < c.length; ++i) {
+    c[i] = 0.5;
+  }
+
+  var source = context.createBufferSource();
+  source.buffer = long_signal;
+  source.connect(merger, 0, 1);
+  source.start(start_offset / buffer_rate);
+
+  return context.startRendering().
+    then((buffer) => {
+      series_output = buffer.getChannelData(0);
+      expected = buffer.getChannelData(1);
+
+      for (var i = 0; i < buffer.length; ++i) {
+        assert_approx_equals(series_output[i], expected[i], EPSILON,
+                             "series output at " + i);
+      }
+    });
+}
+
+function start_tests(context_rate, buffer_rate) {
+  OFFSETS.forEach((start_offset) => {
+    promise_test(() => do_test(context_rate, buffer_rate, start_offset),
+                 "" + context_rate + " context, "
+                 + buffer_rate + " buffer, "
+                 + start_offset + " start");
+  });
+}
+
+start_tests(48000, 44100);
+start_tests(44100, 48000);
+
+</script>
diff --git a/dom/media/webaudio/test/test_singleSourceDest.html b/dom/media/webaudio/test/test_singleSourceDest.html
new file mode 100644
index 0000000000..8613a2dd9d
--- /dev/null
+++ b/dom/media/webaudio/test/test_singleSourceDest.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+  is(destination.context, context, "Destination node has proper context");
+  is(destination.context, context, "Destination node has proper context");
+  is(destination.numberOfInputs, 1, "Destination node has 1 inputs");
+  is(destination.numberOfOutputs, 0, "Destination node has 0 outputs");
+  is(destination.channelCount, 2, "Destination node has 2 input channels by default");
+  is(destination.channelCountMode, "explicit", "Correct channelCountMode for the destination node");
+  is(destination.channelInterpretation, "speakers", "Correct channelCountInterpretation for the destination node");
+  ok(destination instanceof EventTarget, "AudioNodes must be EventTargets");
+
+  var source = context.createBufferSource();
+  is(source.context, context, "Source node has proper context");
+  is(source.numberOfInputs, 0, "Source node has 0 inputs");
+  is(source.numberOfOutputs, 1, "Source node has 1 outputs");
+  is(source.loop, false, "Source node is not looping");
+  is(source.loopStart, 0, "Correct default value for loopStart");
+  is(source.loopEnd, 0, "Correct default value for loopEnd");
+  ok(!source.buffer, "Source node should not have a buffer when it's created");
+  is(source.channelCount, 2, "source node has 2 input channels by default");
+  is(source.channelCountMode, "max", "Correct channelCountMode for the source node");
+  is(source.channelInterpretation, "speakers", "Correct channelCountInterpretation for the source node");
+
+  expectException(function() {
+    source.channelCount = 0;
+  }, DOMException.NOT_SUPPORTED_ERR);
+
+  source.buffer = buffer;
+  ok(source.buffer, "Source node should have a buffer now");
+
+  source.connect(destination);
+
+  is(source.numberOfInputs, 0, "Source node has 0 inputs");
+  is(source.numberOfOutputs, 1, "Source node has 0 outputs");
+  is(destination.numberOfInputs, 1, "Destination node has 0 inputs");
+  is(destination.numberOfOutputs, 0, "Destination node has 0 outputs");
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+
+    SpecialPowers.clearUserPref("media.webaudio.enabled");
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPannerNode.html b/dom/media/webaudio/test/test_stereoPannerNode.html
new file mode 100644
index 0000000000..ffc735364a
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPannerNode.html
@@ -0,0 +1,263 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test StereoPannerNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var SR = 44100;
+var BUF_SIZE = 128;
+var PANNING = 0.1;
+var GAIN = 0.5;
+
+// Cheap reimplementation of some bits of the spec
+function gainForPanningMonoToStereo(panning) {
+  panning += 1;
+  panning /= 2;
+  return [ Math.cos(0.5 * Math.PI * panning),
+           Math.sin(0.5 * Math.PI * panning) ];
+}
+
+function gainForPanningStereoToStereo(panning) {
+  if (panning <= 0) {
+    panning += 1.;
+  }
+  return [ Math.cos(0.5 * Math.PI * panning),
+           Math.sin(0.5 * Math.PI * panning) ];
+}
+
+function applyStereoToStereoPanning(l, r, panningValues, panning) {
+  var outL, outR;
+  if (panning <= 0) {
+    outL = l + r * panningValues[0];
+    outR = r * panningValues[1];
+  } else {
+    outL = l * panningValues[0];
+    outR = r + l * panningValues[1];
+  }
+  return [outL,outR];
+}
+
+function applyMonoToStereoPanning(c, panning) {
+  return [c * panning[0], c * panning[1]];
+}
+
+// Test the DOM interface
+var context = new OfflineAudioContext(1, 1, SR);
+var stereoPanner = context.createStereoPanner();
+ok(stereoPanner.pan, "The AudioParam member must exist");
+is(stereoPanner.pan.value, 0.0, "Correct initial value");
+is(stereoPanner.pan.defaultValue, 0.0, "Correct default value");
+is(stereoPanner.channelCount, 2, "StereoPannerNode node has 2 input channels by default");
+is(stereoPanner.channelCountMode, "clamped-max", "Correct channelCountMode for the StereoPannerNode");
+is(stereoPanner.channelInterpretation, "speakers", "Correct channelCountInterpretation for the StereoPannerNode");
+expectException(function() {
+  stereoPanner.channelCount = 3;
+}, DOMException.NOT_SUPPORTED_ERR);
+expectException(function() {
+  stereoPanner.channelCountMode = "max";
+}, DOMException.NOT_SUPPORTED_ERR);
+
+// A sine to be used to fill the buffers
+function sine(t) {
+  return Math.sin(440 * 2 * Math.PI * t / context.sampleRate);
+}
+
+// A couple mono and stereo buffers: the StereoPannerNode equation is different
+// if the input is mono or stereo
+var stereoBuffer = context.createBuffer(2, BUF_SIZE, context.sampleRate);
+var monoBuffer = context.createBuffer(1, BUF_SIZE, context.sampleRate);
+for (var i = 0; i < BUF_SIZE; ++i) {
+  monoBuffer.getChannelData(0)[i]   =
+  stereoBuffer.getChannelData(0)[i] =
+  stereoBuffer.getChannelData(1)[i] = sine(i);
+}
+
+// Expected test vectors
+function expectedBufferNoop(gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = context.createBuffer(2, BUF_SIZE, SR);
+  for (var i = 0; i < BUF_SIZE; i++) {
+    expectedBuffer.getChannelData(0)[i] = gain * sine(i);
+    expectedBuffer.getChannelData(1)[i] = gain * sine(i);
+  }
+  return expectedBuffer;
+}
+
+function expectedBufferForStereo(gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = context.createBuffer(2, BUF_SIZE, SR);
+  var gainPanning = gainForPanningStereoToStereo(PANNING);
+  gainPanning[0] *= gain;
+  gainPanning[1] *= gain;
+  for (var i = 0; i < BUF_SIZE; i++) {
+    var values = [ sine(i), sine(i) ];
+    var processed = applyStereoToStereoPanning(values[0], values[1], gainPanning, PANNING);
+    expectedBuffer.getChannelData(0)[i] = processed[0];
+    expectedBuffer.getChannelData(1)[i] = processed[1];
+  }
+  return expectedBuffer;
+}
+
+function expectedBufferForMono(gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = context.createBuffer(2, BUF_SIZE, SR);
+  var gainPanning = gainForPanningMonoToStereo(PANNING);
+  gainPanning[0] *= gain;
+  gainPanning[1] *= gain;
+  for (var i = 0; i < BUF_SIZE; i++) {
+    var value = sine(i);
+    var processed = applyMonoToStereoPanning(value, gainPanning);
+    expectedBuffer.getChannelData(0)[i] = processed[0];
+    expectedBuffer.getChannelData(1)[i] = processed[1];
+  }
+  return expectedBuffer;
+}
+
+// Actual test cases
+var tests = [
+  function monoPanningNoop(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    monoSource.start(0);
+    return expectedBufferNoop();
+  },
+  function stereoPanningNoop(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.start(0);
+    return expectedBufferNoop();
+  },
+  function monoPanningNoopWithGain(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    monoSource.start(0);
+    return expectedBufferNoop(GAIN);
+  },
+  function stereoPanningNoopWithGain(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.start(0);
+    return expectedBufferNoop(GAIN);
+  },
+  function stereoPanningAutomation(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.setValueAtTime(0.1, 0.0);
+    stereoSource.start(0);
+    return expectedBufferForStereo();
+  },
+  function stereoPanning(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.connect(panner);
+    panner.pan.value = 0.1;
+    stereoSource.start(0);
+    return expectedBufferForStereo();
+  },
+  function monoPanningAutomation(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0.0);
+    monoSource.start(0);
+    return expectedBufferForMono();
+  },
+  function monoPanning(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.value = 0.1;
+    monoSource.start(0);
+    return expectedBufferForMono();
+  },
+  function monoPanningWithGain(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.value = 0.1;
+    monoSource.start(0);
+    return expectedBufferForMono(GAIN);
+  },
+  function stereoPanningWithGain(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.value = 0.1;
+    stereoSource.start(0);
+    return expectedBufferForStereo(GAIN);
+  },
+  function monoPanningWithGainAndAutomation(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0);
+    monoSource.start(0);
+    return expectedBufferForMono(GAIN);
+  },
+  function stereoPanningWithGainAndAutomation(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0);
+    stereoSource.start(0);
+    return expectedBufferForStereo(GAIN);
+  }
+];
+
+var finished = 0;
+function finish() {
+  if (++finished == tests.length) {
+    SimpleTest.finish();
+  }
+}
+
+tests.forEach(function(f) {
+  var ac = new OfflineAudioContext(2, BUF_SIZE, SR);
+  var panner = ac.createStereoPanner();
+  panner.connect(ac.destination);
+  var expected = f(ac, panner);
+  ac.oncomplete = function(e) {
+    info(f.name);
+    compareBuffers(e.renderedBuffer, expected);
+    finish();
+  };
+  ac.startRendering()
+});
+
+SimpleTest.waitForExplicitFinish();
+
+</script>
+</pre>
+<pre id=dump>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html b/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html
new file mode 100644
index 0000000000..250a1a9de7
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test StereoPanerNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+
+    var stereoPanner = context.createStereoPanner();
+
+    source.buffer = this.buffer;
+
+    source.connect(stereoPanner);
+
+    var stereoPannerWrapped = SpecialPowers.wrap(stereoPanner);
+    ok("passThrough" in stereoPannerWrapped, "StereoPannerNode should support the passThrough API");
+    stereoPannerWrapped.passThrough = true;
+
+    source.start(0);
+    return stereoPanner;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPanningWithGain.html b/dom/media/webaudio/test/test_stereoPanningWithGain.html
new file mode 100644
index 0000000000..1ef0c037df
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPanningWithGain.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test stereo equalpower panning with a GainNode</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const size = 256;
+
+var gTest = {
+  numberOfChannels: 2,
+  createGraph: function(context) {
+    var panner = context.createPanner();
+    panner.setPosition(1.0, 0.0, 0.0); // reference distance the right
+    panner.panningModel = "equalpower";
+
+    var gain = context.createGain();
+    gain.gain.value = -0.5;
+    gain.connect(panner);
+
+    var buffer = context.createBuffer(2, 2, context.sampleRate);
+    buffer.getChannelData(0)[0] = 1.0;
+    buffer.getChannelData(1)[1] = 1.0;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(gain);
+    source.start(0);
+
+    return panner;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(2, size, context.sampleRate);
+    expectedBuffer.getChannelData(1)[0] = -0.5;
+    expectedBuffer.getChannelData(1)[1] = -0.5;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveDecoder.html b/dom/media/webaudio/test/test_waveDecoder.html
new file mode 100644
index 0000000000..bd5faf6adf
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveDecoder.html
@@ -0,0 +1,69 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we decode uint8 and sint16 wave files with correct conversion to float64</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var testsDone = 0;
+var tests = ["UklGRjUrAABXQVZFZm10IBAAAAABAAEAESsAABErAAABAAgAZGF0YQMAAAD/AIA=",
+             "UklGRkZWAABXQVZFZm10IBAAAAABAAEAESsAACJWAAACABAAZGF0YQYAAAD/fwCAAAA="];
+
+SimpleTest.waitForExplicitFinish();
+
+function base64ToUint8Buffer(b64) {
+  var str = atob(b64)
+  var u8 = new Uint8Array(str.length);
+  for (var i = 0; i < str.length; ++i) {
+    u8[i] = str.charCodeAt(i);
+  }
+  return u8;
+}
+
+function fixupBufferSampleRate(u8, rate) {
+  u8[24] = (rate & 0x000000ff) >> 0;
+  u8[25] = (rate & 0x0000ff00) >> 8;
+  u8[26] = (rate & 0x00ff0000) >> 16;
+  u8[27] = (rate & 0xff000000) >> 24;
+}
+
+function finishTest() {
+  testsDone += 1;
+  if (testsDone == tests.length) {
+    SimpleTest.finish();
+  }
+}
+
+function decodeComplete(b) {
+  ok(true, "Decoding succeeded.");
+  is(b.numberOfChannels, 1, "Should have 1 channel.");
+  is(b.length, 3, "Should have three samples.");
+  var samples = b.getChannelData(0);
+  ok(samples[0] >  0.99 && samples[0] <  1.01, "Check near  1.0. Got " + samples[0]);
+  ok(samples[1] > -1.01 && samples[1] < -0.99, "Check near -1.0. Got " + samples[1]);
+  ok(samples[2] > -0.01 && samples[2] <  0.01, "Check near  0.0. Got " + samples[2]);
+  finishTest();
+}
+
+function decodeFailed() {
+  ok(false, "Decoding failed.");
+  finishTest();
+}
+
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  for (var i = 0; i < tests.length; ++i) {
+    var u8 = base64ToUint8Buffer(tests[i]);
+    fixupBufferSampleRate(u8, context.sampleRate);
+    context.decodeAudioData(u8.buffer, decodeComplete, decodeFailed);
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaper.html b/dom/media/webaudio/test/test_waveShaper.html
new file mode 100644
index 0000000000..c95cf5e054
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaper.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+    shaper.curve = this.curve;
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      this.buffer.getChannelData(0)[i] = 2 * (i / 4096) - 1;
+    }
+    // Two out of range values
+    this.buffer.getChannelData(0)[0] = -2;
+    this.buffer.getChannelData(0)[4095] = 2;
+
+    this.curve = new Float32Array(2048);
+    for (var i = 0; i < 2048; ++i) {
+      this.curve[i] = Math.sin(100 * Math.PI * (i + 1) / context.sampleRate);
+    }
+
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      var input = this.buffer.getChannelData(0)[i];
+      var index = Math.floor(this.curve.length * (input + 1) / 2);
+      index = Math.max(0, Math.min(this.curve.length - 1, index));
+      expectedBuffer.getChannelData(0)[i] = this.curve[index];
+    }
+    expectedBuffer.getChannelData(0)[0] = this.curve[0];
+    expectedBuffer.getChannelData(0)[4095] = this.curve[2047];
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperGain.html b/dom/media/webaudio/test/test_waveShaperGain.html
new file mode 100644
index 0000000000..b0c82d2d8c
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperGain.html
@@ -0,0 +1,73 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<meta charset="utf-8">
+  <title>Test that WaveShaperNode doesn't corrupt its inputs when the gain is !=
+    1.0 (bug 1203616)</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+var samplerate = 44100;
+var context = new OfflineAudioContext(1, 44100, samplerate);
+
+var dc = context.createBufferSource();
+
+var buffer = context.createBuffer(1, 1, samplerate);
+buffer.getChannelData(0)[0] = 1.0;
+dc.buffer = buffer;
+
+var gain = context.createGain();
+var ws2 = context.createWaveShaper();
+var ws = [];
+
+// No-op waveshaper curves.
+for (var i = 0; i < 2; i++) {
+  ws[i] = context.createWaveShaper();
+  var curve = new Float32Array(2);
+  curve[0] = -1.0;
+  curve[1] = 1.0;
+  ws[i].curve = curve;
+  ws[i].connect(context.destination);
+  gain.connect(ws[i]);
+}
+
+dc.connect(gain);
+dc.start();
+
+gain.gain.value = 0.5;
+
+context.startRendering().then(buffer => {
+  document.querySelector("pre").innerHTML = buffer.getChannelData(0)[0];
+  ok(buffer.getChannelData(0)[0] == 1.0, "Volume was handled properly");
+
+  context = new OfflineAudioContext(1, 100, samplerate);
+  var oscillator = context.createOscillator();
+  var gain = context.createGain();
+  var waveShaper = context.createWaveShaper();
+
+  oscillator.start(0);
+  oscillator.connect(gain);
+
+  // to silence
+  gain.gain.value = 0;
+  gain.connect(waveShaper);
+
+  // convert all signal into 1.0. The non unity values are to detect the use
+  // of uninitialized buffers (see Bug 1283910).
+  waveShaper.curve = new Float32Array([ 0.5, 0.5, 0.5, 0.5, 0.5, 1, 1, 0.5, 0.5, 0.5, 0.5, 0.5 ]);
+  waveShaper.connect(context.destination);
+
+  context.startRendering().then((buffer) => {
+    var result = buffer.getChannelData(0);
+    ok(result.every(x => x === 1), "WaveShaper handles zero gain properly");
+    SimpleTest.finish();
+  });
+});
+</script>
+<pre>
+</pre>
+</body>
+
diff --git a/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html b/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html
new file mode 100644
index 0000000000..f117f03764
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with an invalid curve</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+
+    expectException(() => {
+        shaper.curve = new Float32Array(0);
+    }, DOMException.INVALID_STATE_ERR);
+
+    is(shaper.curve, null, "The curve mustn't have been set");
+
+    expectException(() => {
+        shaper.curve = new Float32Array(1);
+    }, DOMException.INVALID_STATE_ERR);
+
+    is(shaper.curve, null, "The curve mustn't have been set");
+
+    expectNoException(() => {
+        shaper.curve = new Float32Array(2);
+    });
+
+    isnot(shaper.curve, null, "The curve must have been set");
+
+    expectNoException(() => {
+        shaper.curve = null;
+    });
+
+    is(shaper.curve, null, "The curve must be null by default");
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperNoCurve.html b/dom/media/webaudio/test/test_waveShaperNoCurve.html
new file mode 100644
index 0000000000..c0d3187b2c
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperNoCurve.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+    is(shaper.curve, null, "The shaper curve must be null by default");
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers: function(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperPassThrough.html b/dom/media/webaudio/test/test_waveShaperPassThrough.html
new file mode 100644
index 0000000000..52c70d3c2f
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperPassThrough.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with passthrough</title>
+  <script type="text/javascript" src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph: function(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+    shaper.curve = this.curve;
+
+    var shaperWrapped = SpecialPowers.wrap(shaper);
+    ok("passThrough" in shaperWrapped, "WaveShaperNode should support the passThrough API");
+    shaperWrapped.passThrough = true;
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers: function(context) {
+    this.buffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      this.buffer.getChannelData(0)[i] = 2 * (i / 4096) - 1;
+    }
+    // Two out of range values
+    this.buffer.getChannelData(0)[0] = -2;
+    this.buffer.getChannelData(0)[4095] = 2;
+
+    this.curve = new Float32Array(2048);
+    for (var i = 0; i < 2048; ++i) {
+      this.curve[i] = Math.sin(100 * Math.PI * (i + 1) / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/ting-44.1k-1ch.ogg b/dom/media/webaudio/test/ting-44.1k-1ch.ogg
new file mode 100644
index 0000000000..a11aaf1cbf
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-1ch.ogg
diff --git a/dom/media/webaudio/test/ting-44.1k-1ch.wav b/dom/media/webaudio/test/ting-44.1k-1ch.wav
new file mode 100644
index 0000000000..6854c9d898
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-1ch.wav
diff --git a/dom/media/webaudio/test/ting-44.1k-2ch.ogg b/dom/media/webaudio/test/ting-44.1k-2ch.ogg
new file mode 100644
index 0000000000..94e0014858
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-2ch.ogg
diff --git a/dom/media/webaudio/test/ting-44.1k-2ch.wav b/dom/media/webaudio/test/ting-44.1k-2ch.wav
new file mode 100644
index 0000000000..703d885892
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-2ch.wav
diff --git a/dom/media/webaudio/test/ting-48k-1ch.ogg b/dom/media/webaudio/test/ting-48k-1ch.ogg
new file mode 100644
index 0000000000..f45ce33a58
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-1ch.ogg
diff --git a/dom/media/webaudio/test/ting-48k-1ch.wav b/dom/media/webaudio/test/ting-48k-1ch.wav
new file mode 100644
index 0000000000..8fe471666c
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-1ch.wav
diff --git a/dom/media/webaudio/test/ting-48k-2ch.ogg b/dom/media/webaudio/test/ting-48k-2ch.ogg
new file mode 100644
index 0000000000..e4c564abbd
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-2ch.ogg
diff --git a/dom/media/webaudio/test/ting-48k-2ch.wav b/dom/media/webaudio/test/ting-48k-2ch.wav
new file mode 100644
index 0000000000..ad4d0466da
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-2ch.wav
diff --git a/dom/media/webaudio/test/ting-dualchannel44.1.wav b/dom/media/webaudio/test/ting-dualchannel44.1.wav
new file mode 100644
index 0000000000..62954394d3
--- /dev/null
+++ b/dom/media/webaudio/test/ting-dualchannel44.1.wav
diff --git a/dom/media/webaudio/test/ting-dualchannel48.wav b/dom/media/webaudio/test/ting-dualchannel48.wav
new file mode 100644
index 0000000000..a0b8247888
--- /dev/null
+++ b/dom/media/webaudio/test/ting-dualchannel48.wav
diff --git a/dom/media/webaudio/test/webaudio.js b/dom/media/webaudio/test/webaudio.js
new file mode 100644
index 0000000000..1a1a8efb74
--- /dev/null
+++ b/dom/media/webaudio/test/webaudio.js
@@ -0,0 +1,269 @@
+// Helpers for Web Audio tests
+
+function expectException(func, exceptionCode) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+    ok(ex instanceof DOMException, "Expect a DOM exception");
+    is(ex.code, exceptionCode, "Expect the correct exception code");
+  }
+  ok(threw, "The exception was thrown");
+}
+
+function expectNoException(func) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+  }
+  ok(!threw, "An exception was not thrown");
+}
+
+function expectTypeError(func) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+    ok(ex instanceof TypeError, "Expect a TypeError");
+  }
+  ok(threw, "The exception was thrown");
+}
+
+function expectRejectedPromise(that, func, exceptionName) {
+  var promise = that[func]();
+
+  ok(promise instanceof Promise, "Expect a Promise");
+
+  promise.then(function(res) {
+    ok(false, "Promise resolved when it should have been rejected.");
+  }).catch(function(err) {
+    is(err.name, exceptionName, "Promise correctly reject with " + exceptionName);
+  });
+}
+
+function fuzzyCompare(a, b) {
+  return Math.abs(a - b) < 9e-3;
+}
+
+function compareChannels(buf1, buf2,
+                        /*optional*/ length,
+                        /*optional*/ sourceOffset,
+                        /*optional*/ destOffset,
+                        /*optional*/ skipLengthCheck) {
+  if (!skipLengthCheck) {
+    is(buf1.length, buf2.length, "Channels must have the same length");
+  }
+  sourceOffset = sourceOffset || 0;
+  destOffset = destOffset || 0;
+  if (length == undefined) {
+    length = buf1.length - sourceOffset;
+  }
+  var difference = 0;
+  var maxDifference = 0;
+  var firstBadIndex = -1;
+  for (var i = 0; i < length; ++i) {
+    if (!fuzzyCompare(buf1[i + sourceOffset], buf2[i + destOffset])) {
+      difference++;
+      maxDifference = Math.max(maxDifference, Math.abs(buf1[i + sourceOffset] - buf2[i + destOffset]));
+      if (firstBadIndex == -1) {
+        firstBadIndex = i;
+      }
+    }
+  };
+
+  is(difference, 0, "maxDifference: " + maxDifference +
+     ", first bad index: " + firstBadIndex +
+     " with test-data offset " + sourceOffset + " and expected-data offset " +
+     destOffset + "; corresponding values " + buf1[firstBadIndex + sourceOffset] +
+     " and " + buf2[firstBadIndex + destOffset] + " --- differences");
+}
+
+function compareBuffers(got, expected) {
+  if (got.numberOfChannels != expected.numberOfChannels) {
+    is(got.numberOfChannels, expected.numberOfChannels,
+       "Correct number of buffer channels");
+    return;
+  }
+  if (got.length != expected.length) {
+    is(got.length, expected.length,
+       "Correct buffer length");
+    return;
+  }
+  if (got.sampleRate != expected.sampleRate) {
+    is(got.sampleRate, expected.sampleRate,
+       "Correct sample rate");
+    return;
+  }
+
+  for (var i = 0; i < got.numberOfChannels; ++i) {
+    compareChannels(got.getChannelData(i), expected.getChannelData(i),
+                    got.length, 0, 0, true);
+  }
+}
+
+/**
+ * Compute the root mean square (RMS,
+ * <http://en.wikipedia.org/wiki/Root_mean_square>) of a channel of a slice
+ * (defined by `start` and `end`) of an AudioBuffer.
+ *
+ * This is useful to detect that a buffer is noisy or silent.
+ */
+function rms(audiobuffer, channel = 0, start = 0, end = audiobuffer.length) {
+  var buffer= audiobuffer.getChannelData(channel);
+  var rms = 0;
+  for (var i = start; i < end; i++) {
+    rms += buffer[i] * buffer[i];
+  }
+
+  rms /= buffer.length;
+  rms = Math.sqrt(rms);
+  return rms;
+}
+
+function getEmptyBuffer(context, length) {
+  return context.createBuffer(gTest.numberOfChannels, length, context.sampleRate);
+}
+
+/**
+ * This function assumes that the test file defines a single gTest variable with
+ * the following properties and methods:
+ *
+ * + numberOfChannels: optional property which specifies the number of channels
+ *                     in the output.  The default value is 2.
+ * + createGraph: mandatory method which takes a context object and does
+ *                everything needed in order to set up the Web Audio graph.
+ *                This function returns the node to be inspected.
+ * + createGraphAsync: async version of createGraph.  This function takes
+ *                     a callback which should be called with an argument
+ *                     set to the node to be inspected when the callee is
+ *                     ready to proceed with the test.  Either this function
+ *                     or createGraph must be provided.
+ * + createExpectedBuffers: optional method which takes a context object and
+ *                          returns either one expected buffer or an array of
+ *                          them, designating what is expected to be observed
+ *                          in the output.  If omitted, the output is expected
+ *                          to be silence.  All buffers must have the same
+ *                          length, which must be a bufferSize supported by
+ *                          ScriptProcessorNode.  This function is guaranteed
+ *                          to be called before createGraph.
+ * + length: property equal to the total number of frames which we are waiting
+ *           to see in the output, mandatory if createExpectedBuffers is not
+ *           provided, in which case it must be a bufferSize supported by
+ *           ScriptProcessorNode (256, 512, 1024, 2048, 4096, 8192, or 16384).
+ *           If createExpectedBuffers is provided then this must be equal to
+ *           the number of expected buffers * the expected buffer length.
+ *
+ * + skipOfflineContextTests: optional. when true, skips running tests on an offline
+ *                            context by circumventing testOnOfflineContext.
+ */
+function runTest()
+{
+  function done() {
+    SimpleTest.finish();
+  }
+
+  SimpleTest.waitForExplicitFinish();
+  function runTestFunction () {
+    if (!gTest.numberOfChannels) {
+      gTest.numberOfChannels = 2; // default
+    }
+
+    var testLength;
+
+    function runTestOnContext(context, callback, testOutput) {
+      if (!gTest.createExpectedBuffers) {
+        // Assume that the output is silence
+        var expectedBuffers = getEmptyBuffer(context, gTest.length);
+      } else {
+        var expectedBuffers = gTest.createExpectedBuffers(context);
+      }
+      if (!(expectedBuffers instanceof Array)) {
+        expectedBuffers = [expectedBuffers];
+      }
+      var expectedFrames = 0;
+      for (var i = 0; i < expectedBuffers.length; ++i) {
+        is(expectedBuffers[i].numberOfChannels, gTest.numberOfChannels,
+           "Correct number of channels for expected buffer " + i);
+        expectedFrames += expectedBuffers[i].length;
+      }
+      if (gTest.length && gTest.createExpectedBuffers) {
+        is(expectedFrames, gTest.length, "Correct number of expected frames");
+      }
+
+      if (gTest.createGraphAsync) {
+        gTest.createGraphAsync(context, function(nodeToInspect) {
+          testOutput(nodeToInspect, expectedBuffers, callback);
+        });
+      } else {
+        testOutput(gTest.createGraph(context), expectedBuffers, callback);
+      }
+    }
+
+    function testOnNormalContext(callback) {
+      function testOutput(nodeToInspect, expectedBuffers, callback) {
+        testLength = 0;
+        var sp = context.createScriptProcessor(expectedBuffers[0].length, gTest.numberOfChannels, 0);
+        nodeToInspect.connect(sp);
+        sp.onaudioprocess = function(e) {
+          var expectedBuffer = expectedBuffers.shift();
+          testLength += expectedBuffer.length;
+          compareBuffers(e.inputBuffer, expectedBuffer);
+          if (expectedBuffers.length == 0) {
+            sp.onaudioprocess = null;
+            callback();
+          }
+        };
+      }
+      var context = new AudioContext();
+      runTestOnContext(context, callback, testOutput);
+    }
+
+    function testOnOfflineContext(callback, sampleRate) {
+      function testOutput(nodeToInspect, expectedBuffers, callback) {
+        nodeToInspect.connect(context.destination);
+        context.oncomplete = function(e) {
+          var samplesSeen = 0;
+          while (expectedBuffers.length) {
+            var expectedBuffer = expectedBuffers.shift();
+            is(e.renderedBuffer.numberOfChannels, expectedBuffer.numberOfChannels,
+               "Correct number of input buffer channels");
+            for (var i = 0; i < e.renderedBuffer.numberOfChannels; ++i) {
+              compareChannels(e.renderedBuffer.getChannelData(i),
+                             expectedBuffer.getChannelData(i),
+                             expectedBuffer.length,
+                             samplesSeen,
+                             undefined,
+                             true);
+            }
+            samplesSeen += expectedBuffer.length;
+          }
+          callback();
+        };
+        context.startRendering();
+      }
+
+      var context = new OfflineAudioContext(gTest.numberOfChannels, testLength, sampleRate);
+      runTestOnContext(context, callback, testOutput);
+    }
+
+    testOnNormalContext(function() {
+      if (!gTest.skipOfflineContextTests) {
+        testOnOfflineContext(function() {
+          testOnOfflineContext(done, 44100);
+        }, 48000);
+      } else {
+        done();
+      }
+    });
+  };
+
+  if (document.readyState !== 'complete') {
+    addLoadEvent(runTestFunction);
+  } else {
+    runTestFunction();
+  }
+}