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122 lines
3.9 KiB
C++
122 lines
3.9 KiB
C++
/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim: set ts=8 sts=2 et sw=2 tw=80: */
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#if !defined(AudioCompactor_h)
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#define AudioCompactor_h
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#include "MediaQueue.h"
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#include "MediaData.h"
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#include "VideoUtils.h"
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namespace mozilla {
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class AudioCompactor
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{
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public:
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AudioCompactor(MediaQueue<AudioData>& aQueue)
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: mQueue(aQueue)
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{ }
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// Push audio data into the underlying queue with minimal heap allocation
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// slop. This method is responsible for allocating AudioDataValue[] buffers.
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// The caller must provide a functor to copy the data into the buffers. The
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// functor must provide the following signature:
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//
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// uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
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//
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// The functor must copy as many complete frames as possible to the provided
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// buffer given its length (in AudioDataValue elements). The number of frames
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// copied must be returned. This copy functor must support being called
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// multiple times in order to copy the audio data fully. The copy functor
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// must copy full frames as partial frames will be ignored.
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template<typename CopyFunc>
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bool Push(int64_t aOffset, int64_t aTime, int32_t aSampleRate,
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uint32_t aFrames, uint32_t aChannels, CopyFunc aCopyFunc)
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{
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// If we are losing more than a reasonable amount to padding, try to chunk
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// the data.
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size_t maxSlop = AudioDataSize(aFrames, aChannels) / MAX_SLOP_DIVISOR;
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while (aFrames > 0) {
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uint32_t samples = GetChunkSamples(aFrames, aChannels, maxSlop);
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nsAutoArrayPtr<AudioDataValue> buffer(new AudioDataValue[samples]);
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// Copy audio data to buffer using caller-provided functor.
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uint32_t framesCopied = aCopyFunc(buffer, samples);
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NS_ASSERTION(framesCopied <= aFrames, "functor copied too many frames");
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CheckedInt64 duration = FramesToUsecs(framesCopied, aSampleRate);
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if (!duration.isValid()) {
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return false;
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}
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mQueue.Push(new AudioData(aOffset,
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aTime,
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duration.value(),
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framesCopied,
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buffer.forget(),
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aChannels));
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// Remove the frames we just pushed into the queue and loop if there is
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// more to be done.
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aTime += duration.value();
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aFrames -= framesCopied;
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// NOTE: No need to update aOffset as its only an approximation anyway.
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}
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return true;
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}
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// Copy functor suitable for copying audio samples already in the
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// AudioDataValue format/layout expected by AudioStream on this platform.
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class NativeCopy
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{
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public:
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NativeCopy(const uint8_t* aSource, size_t aSourceBytes,
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uint32_t aChannels)
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: mSource(aSource)
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, mSourceBytes(aSourceBytes)
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, mChannels(aChannels)
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, mNextByte(0)
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{ }
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uint32_t operator()(AudioDataValue *aBuffer, uint32_t aSamples);
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private:
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const uint8_t* const mSource;
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const size_t mSourceBytes;
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const uint32_t mChannels;
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size_t mNextByte;
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};
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// Allow 12.5% slop before chunking kicks in. Public so that the gtest can
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// access it.
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static const size_t MAX_SLOP_DIVISOR = 8;
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private:
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// Compute the number of AudioDataValue samples that will be fit the most
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// frames while keeping heap allocation slop less than the given threshold.
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static uint32_t
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GetChunkSamples(uint32_t aFrames, uint32_t aChannels, size_t aMaxSlop);
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static size_t BytesPerFrame(uint32_t aChannels)
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{
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return sizeof(AudioDataValue) * aChannels;
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}
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static size_t AudioDataSize(uint32_t aFrames, uint32_t aChannels)
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{
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return aFrames * BytesPerFrame(aChannels);
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}
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MediaQueue<AudioData> &mQueue;
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};
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} // namespace mozilla
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#endif // AudioCompactor_h
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