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Diffstat (limited to 'dom/media/AudioStream.h')
-rw-r--r-- | dom/media/AudioStream.h | 308 |
1 files changed, 308 insertions, 0 deletions
diff --git a/dom/media/AudioStream.h b/dom/media/AudioStream.h new file mode 100644 index 0000000000..acc38b93db --- /dev/null +++ b/dom/media/AudioStream.h @@ -0,0 +1,308 @@ +/* -*- 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/. */ +#if !defined(AudioStream_h_) +#define AudioStream_h_ + +#include "AudioSampleFormat.h" +#include "nsAutoPtr.h" +#include "nsCOMPtr.h" +#include "nsThreadUtils.h" +#include "mozilla/dom/AudioChannelBinding.h" +#include "mozilla/Monitor.h" +#include "mozilla/RefPtr.h" +#include "mozilla/TimeStamp.h" +#include "mozilla/UniquePtr.h" +#include "CubebUtils.h" +#include "soundtouch/SoundTouchFactory.h" + +namespace mozilla { + +struct CubebDestroyPolicy +{ + void operator()(cubeb_stream* aStream) const { + cubeb_stream_destroy(aStream); + } +}; + +class AudioStream; +class FrameHistory; +class AudioConfig; +class AudioConverter; + +class AudioClock +{ +public: + AudioClock(); + + // Initialize the clock with the current sampling rate. + // Need to be called before querying the clock. + void Init(uint32_t aRate); + + // Update the number of samples that has been written in the audio backend. + // Called on the state machine thread. + void UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun); + + /** + * @param aFrames The playback position in frames of the audio engine. + * @return The playback position in frames of the stream, + * adjusted by playback rate changes and underrun frames. + */ + int64_t GetPositionInFrames(int64_t aFrames) const; + + /** + * @param frames The playback position in frames of the audio engine. + * @return The playback position in microseconds of the stream, + * adjusted by playback rate changes and underrun frames. + */ + int64_t GetPosition(int64_t frames) const; + + // Set the playback rate. + // Called on the audio thread. + void SetPlaybackRate(double aPlaybackRate); + // Get the current playback rate. + // Called on the audio thread. + double GetPlaybackRate() const; + // Set if we are preserving the pitch. + // Called on the audio thread. + void SetPreservesPitch(bool aPreservesPitch); + // Get the current pitch preservation state. + // Called on the audio thread. + bool GetPreservesPitch() const; + + uint32_t GetInputRate() const { return mInRate; } + uint32_t GetOutputRate() const { return mOutRate; } + +private: + // Output rate in Hz (characteristic of the playback rate) + uint32_t mOutRate; + // Input rate in Hz (characteristic of the media being played) + uint32_t mInRate; + // True if the we are timestretching, false if we are resampling. + bool mPreservesPitch; + // The history of frames sent to the audio engine in each DataCallback. + const nsAutoPtr<FrameHistory> mFrameHistory; +}; + +/* + * A bookkeeping class to track the read/write position of an audio buffer. + */ +class AudioBufferCursor { +public: + AudioBufferCursor(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames) + : mPtr(aPtr), mChannels(aChannels), mFrames(aFrames) {} + + // Advance the cursor to account for frames that are consumed. + uint32_t Advance(uint32_t aFrames) { + MOZ_ASSERT(mFrames >= aFrames); + mFrames -= aFrames; + mPtr += mChannels * aFrames; + return aFrames; + } + + // The number of frames available for read/write in this buffer. + uint32_t Available() const { return mFrames; } + + // Return a pointer where read/write should begin. + AudioDataValue* Ptr() const { return mPtr; } + +protected: + AudioDataValue* mPtr; + const uint32_t mChannels; + uint32_t mFrames; +}; + +/* + * A helper class to encapsulate pointer arithmetic and provide means to modify + * the underlying audio buffer. + */ +class AudioBufferWriter : private AudioBufferCursor { +public: + AudioBufferWriter(AudioDataValue* aPtr, uint32_t aChannels, uint32_t aFrames) + : AudioBufferCursor(aPtr, aChannels, aFrames) {} + + uint32_t WriteZeros(uint32_t aFrames) { + memset(mPtr, 0, sizeof(AudioDataValue) * mChannels * aFrames); + return Advance(aFrames); + } + + uint32_t Write(const AudioDataValue* aPtr, uint32_t aFrames) { + memcpy(mPtr, aPtr, sizeof(AudioDataValue) * mChannels * aFrames); + return Advance(aFrames); + } + + // Provide a write fuction to update the audio buffer with the following + // signature: uint32_t(const AudioDataValue* aPtr, uint32_t aFrames) + // aPtr: Pointer to the audio buffer. + // aFrames: The number of frames available in the buffer. + // return: The number of frames actually written by the function. + template <typename Function> + uint32_t Write(const Function& aFunction, uint32_t aFrames) { + return Advance(aFunction(mPtr, aFrames)); + } + + using AudioBufferCursor::Available; +}; + +// Access to a single instance of this class must be synchronized by +// callers, or made from a single thread. One exception is that access to +// GetPosition, GetPositionInFrames, SetVolume, and Get{Rate,Channels}, +// SetMicrophoneActive is thread-safe without external synchronization. +class AudioStream final +{ + virtual ~AudioStream(); + +public: + NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream) + + class Chunk { + public: + // Return a pointer to the audio data. + virtual const AudioDataValue* Data() const = 0; + // Return the number of frames in this chunk. + virtual uint32_t Frames() const = 0; + // Return the number of audio channels. + virtual uint32_t Channels() const = 0; + // Return the sample rate of this chunk. + virtual uint32_t Rate() const = 0; + // Return a writable pointer for downmixing. + virtual AudioDataValue* GetWritable() const = 0; + virtual ~Chunk() {} + }; + + class DataSource { + public: + // Return a chunk which contains at most aFrames frames or zero if no + // frames in the source at all. + virtual UniquePtr<Chunk> PopFrames(uint32_t aFrames) = 0; + // Return true if no more data will be added to the source. + virtual bool Ended() const = 0; + // Notify that all data is drained by the AudioStream. + virtual void Drained() = 0; + protected: + virtual ~DataSource() {} + }; + + explicit AudioStream(DataSource& aSource); + + // Initialize the audio stream. aNumChannels is the number of audio + // channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate + // (22050Hz, 44100Hz, etc). + nsresult Init(uint32_t aNumChannels, uint32_t aRate, + const dom::AudioChannel aAudioStreamChannel); + + // Closes the stream. All future use of the stream is an error. + void Shutdown(); + + void Reset(); + + // Set the current volume of the audio playback. This is a value from + // 0 (meaning muted) to 1 (meaning full volume). Thread-safe. + void SetVolume(double aVolume); + + // Start the stream. + void Start(); + + // Pause audio playback. + void Pause(); + + // Resume audio playback. + void Resume(); + + // Return the position in microseconds of the audio frame being played by + // the audio hardware, compensated for playback rate change. Thread-safe. + int64_t GetPosition(); + + // Return the position, measured in audio frames played since the stream + // was opened, of the audio hardware. Thread-safe. + int64_t GetPositionInFrames(); + + static uint32_t GetPreferredRate() + { + return CubebUtils::PreferredSampleRate(); + } + + uint32_t GetOutChannels() { return mOutChannels; } + + // Set playback rate as a multiple of the intrinsic playback rate. This is to + // be called only with aPlaybackRate > 0.0. + nsresult SetPlaybackRate(double aPlaybackRate); + // Switch between resampling (if false) and time stretching (if true, default). + nsresult SetPreservesPitch(bool aPreservesPitch); + + size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const; + +protected: + friend class AudioClock; + + // Return the position, measured in audio frames played since the stream was + // opened, of the audio hardware, not adjusted for the changes of playback + // rate or underrun frames. + // Caller must own the monitor. + int64_t GetPositionInFramesUnlocked(); + +private: + nsresult OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams, + TimeStamp aStartTime, bool aIsFirst); + + static long DataCallback_S(cubeb_stream*, void* aThis, + const void* /* aInputBuffer */, void* aOutputBuffer, + long aFrames) + { + return static_cast<AudioStream*>(aThis)->DataCallback(aOutputBuffer, aFrames); + } + + static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState) + { + static_cast<AudioStream*>(aThis)->StateCallback(aState); + } + + + long DataCallback(void* aBuffer, long aFrames); + void StateCallback(cubeb_state aState); + + nsresult EnsureTimeStretcherInitializedUnlocked(); + + // Return true if audio frames are valid (correct sampling rate and valid + // channel count) otherwise false. + bool IsValidAudioFormat(Chunk* aChunk); + + void GetUnprocessed(AudioBufferWriter& aWriter); + void GetTimeStretched(AudioBufferWriter& aWriter); + + template <typename Function, typename... Args> + int InvokeCubeb(Function aFunction, Args&&... aArgs); + + // The monitor is held to protect all access to member variables. + Monitor mMonitor; + + uint32_t mChannels; + uint32_t mOutChannels; + AudioClock mAudioClock; + soundtouch::SoundTouch* mTimeStretcher; + + // Output file for dumping audio + FILE* mDumpFile; + + // Owning reference to a cubeb_stream. + UniquePtr<cubeb_stream, CubebDestroyPolicy> mCubebStream; + + enum StreamState { + INITIALIZED, // Initialized, playback has not begun. + STARTED, // cubeb started. + STOPPED, // Stopped by a call to Pause(). + DRAINED, // StateCallback has indicated that the drain is complete. + ERRORED, // Stream disabled due to an internal error. + SHUTDOWN // Shutdown has been called + }; + + StreamState mState; + + DataSource& mDataSource; +}; + +} // namespace mozilla + +#endif |