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/* -*- Mode: C++; tab-width: 8; 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/. */
#if !defined(AudioCompactor_h)
#define AudioCompactor_h
#include "MediaQueue.h"
#include "MediaData.h"
#include "VideoUtils.h"
namespace mozilla {
class AudioCompactor
{
public:
explicit AudioCompactor(MediaQueue<AudioData>& aQueue)
: mQueue(aQueue)
{
// Determine padding size used by AlignedBuffer.
size_t paddedSize = AlignedAudioBuffer::AlignmentPaddingSize();
mSamplesPadding = paddedSize / sizeof(AudioDataValue);
if (mSamplesPadding * sizeof(AudioDataValue) < paddedSize) {
// Round up.
mSamplesPadding++;
}
}
// Push audio data into the underlying queue with minimal heap allocation
// slop. This method is responsible for allocating AudioDataValue[] buffers.
// The caller must provide a functor to copy the data into the buffers. The
// functor must provide the following signature:
//
// uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
//
// The functor must copy as many complete frames as possible to the provided
// buffer given its length (in AudioDataValue elements). The number of frames
// copied must be returned. This copy functor must support being called
// multiple times in order to copy the audio data fully. The copy functor
// must copy full frames as partial frames will be ignored.
template<typename CopyFunc>
bool Push(int64_t aOffset, int64_t aTime, int32_t aSampleRate,
uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc)
{
// If we are losing more than a reasonable amount to padding, try to chunk
// the data.
size_t maxSlop = AudioDataSize(aFrames, aChannels) / MAX_SLOP_DIVISOR;
while (aFrames > 0) {
uint32_t samples = GetChunkSamples(aFrames, aChannels, maxSlop);
if (samples / aChannels > mSamplesPadding / aChannels + 1) {
samples -= mSamplesPadding;
}
AlignedAudioBuffer buffer(samples);
if (!buffer) {
return false;
}
// Copy audio data to buffer using caller-provided functor.
uint32_t framesCopied = aCopyFunc(buffer.get(), samples);
NS_ASSERTION(framesCopied <= aFrames, "functor copied too many frames");
buffer.SetLength(size_t(framesCopied) * aChannels);
CheckedInt64 duration = FramesToUsecs(framesCopied, aSampleRate);
if (!duration.isValid()) {
return false;
}
mQueue.Push(new AudioData(aOffset,
aTime,
duration.value(),
framesCopied,
Move(buffer),
aChannels,
aSampleRate));
// Remove the frames we just pushed into the queue and loop if there is
// more to be done.
aTime += duration.value();
aFrames -= framesCopied;
// NOTE: No need to update aOffset as its only an approximation anyway.
}
return true;
}
// Copy functor suitable for copying audio samples already in the
// AudioDataValue format/layout expected by AudioStream on this platform.
class NativeCopy
{
public:
NativeCopy(const uint8_t* aSource, size_t aSourceBytes,
uint32_t aChannels)
: mSource(aSource)
, mSourceBytes(aSourceBytes)
, mChannels(aChannels)
, mNextByte(0)
{ }
uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
private:
const uint8_t* const mSource;
const size_t mSourceBytes;
const uint32_t mChannels;
size_t mNextByte;
};
// Allow 12.5% slop before chunking kicks in. Public so that the gtest can
// access it.
static const size_t MAX_SLOP_DIVISOR = 8;
private:
// Compute the number of AudioDataValue samples that will be fit the most
// frames while keeping heap allocation slop less than the given threshold.
static uint32_t
GetChunkSamples(uint32_t aFrames, uint32_t aChannels, size_t aMaxSlop);
static size_t BytesPerFrame(uint32_t aChannels)
{
return sizeof(AudioDataValue) * aChannels;
}
static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels)
{
return aFrames * BytesPerFrame(aChannels);
}
MediaQueue<AudioData> &mQueue;
size_t mSamplesPadding;
};
} // namespace mozilla
#endif // AudioCompactor_h
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