424 lines
14 KiB
C++
424 lines
14 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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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 file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef MOZILLA_AUDIOSEGMENT_H_
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#define MOZILLA_AUDIOSEGMENT_H_
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#include "MediaSegment.h"
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#include "AudioSampleFormat.h"
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#include "AudioChannelFormat.h"
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#include "SharedBuffer.h"
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#include "WebAudioUtils.h"
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#ifdef MOZILLA_INTERNAL_API
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#include "mozilla/TimeStamp.h"
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#endif
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#include <float.h>
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namespace mozilla {
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template<typename T>
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class SharedChannelArrayBuffer : public ThreadSharedObject {
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public:
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explicit SharedChannelArrayBuffer(nsTArray<nsTArray<T> >* aBuffers)
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{
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mBuffers.SwapElements(*aBuffers);
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}
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virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
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{
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size_t amount = 0;
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amount += mBuffers.ShallowSizeOfExcludingThis(aMallocSizeOf);
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for (size_t i = 0; i < mBuffers.Length(); i++) {
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amount += mBuffers[i].ShallowSizeOfExcludingThis(aMallocSizeOf);
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}
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return amount;
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}
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virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
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{
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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nsTArray<nsTArray<T> > mBuffers;
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};
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class AudioMixer;
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/**
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* For auto-arrays etc, guess this as the common number of channels.
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*/
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const int GUESS_AUDIO_CHANNELS = 2;
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// We ensure that the graph advances in steps that are multiples of the Web
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// Audio block size
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const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
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const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
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template <typename SrcT, typename DestT>
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static void
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InterleaveAndConvertBuffer(const SrcT* const* aSourceChannels,
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uint32_t aLength, float aVolume,
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uint32_t aChannels,
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DestT* aOutput)
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{
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DestT* output = aOutput;
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for (size_t i = 0; i < aLength; ++i) {
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for (size_t channel = 0; channel < aChannels; ++channel) {
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float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
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*output = FloatToAudioSample<DestT>(v);
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++output;
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}
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}
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}
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template <typename SrcT, typename DestT>
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static void
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DeinterleaveAndConvertBuffer(const SrcT* aSourceBuffer,
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uint32_t aFrames, uint32_t aChannels,
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DestT** aOutput)
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{
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for (size_t i = 0; i < aChannels; i++) {
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size_t interleavedIndex = i;
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for (size_t j = 0; j < aFrames; j++) {
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ConvertAudioSample(aSourceBuffer[interleavedIndex],
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aOutput[i][j]);
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interleavedIndex += aChannels;
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}
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}
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}
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class SilentChannel
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{
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public:
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static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
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static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES];
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// We take advantage of the fact that zero in float and zero in int have the
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// same all-zeros bit layout.
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template<typename T>
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static const T* ZeroChannel();
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};
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/**
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* Given an array of input channels (aChannelData), downmix to aOutputChannels,
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* interleave the channel data. A total of aOutputChannels*aDuration
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* interleaved samples will be copied to a channel buffer in aOutput.
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*/
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template <typename SrcT, typename DestT>
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void
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DownmixAndInterleave(const nsTArray<const SrcT*>& aChannelData,
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int32_t aDuration, float aVolume, uint32_t aOutputChannels,
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DestT* aOutput)
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{
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if (aChannelData.Length() == aOutputChannels) {
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InterleaveAndConvertBuffer(aChannelData.Elements(),
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aDuration, aVolume, aOutputChannels, aOutput);
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} else {
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nsAutoTArray<SrcT*,GUESS_AUDIO_CHANNELS> outputChannelData;
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nsAutoTArray<SrcT, SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS> outputBuffers;
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outputChannelData.SetLength(aOutputChannels);
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outputBuffers.SetLength(aDuration * aOutputChannels);
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for (uint32_t i = 0; i < aOutputChannels; i++) {
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outputChannelData[i] = outputBuffers.Elements() + aDuration * i;
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}
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AudioChannelsDownMix(aChannelData,
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outputChannelData.Elements(),
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aOutputChannels,
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aDuration);
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InterleaveAndConvertBuffer(outputChannelData.Elements(),
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aDuration, aVolume, aOutputChannels, aOutput);
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}
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}
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/**
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* An AudioChunk represents a multi-channel buffer of audio samples.
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* It references an underlying ThreadSharedObject which manages the lifetime
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* of the buffer. An AudioChunk maintains its own duration and channel data
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* pointers so it can represent a subinterval of a buffer without copying.
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* An AudioChunk can store its individual channels anywhere; it maintains
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* separate pointers to each channel's buffer.
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*/
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struct AudioChunk {
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typedef mozilla::AudioSampleFormat SampleFormat;
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// Generic methods
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void SliceTo(StreamTime aStart, StreamTime aEnd)
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{
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MOZ_ASSERT(aStart >= 0 && aStart < aEnd && aEnd <= mDuration,
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"Slice out of bounds");
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if (mBuffer) {
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MOZ_ASSERT(aStart < INT32_MAX, "Can't slice beyond 32-bit sample lengths");
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for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
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mChannelData[channel] = AddAudioSampleOffset(mChannelData[channel],
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mBufferFormat, int32_t(aStart));
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}
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}
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mDuration = aEnd - aStart;
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}
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StreamTime GetDuration() const { return mDuration; }
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bool CanCombineWithFollowing(const AudioChunk& aOther) const
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{
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if (aOther.mBuffer != mBuffer) {
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return false;
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}
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if (mBuffer) {
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NS_ASSERTION(aOther.mBufferFormat == mBufferFormat,
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"Wrong metadata about buffer");
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NS_ASSERTION(aOther.mChannelData.Length() == mChannelData.Length(),
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"Mismatched channel count");
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if (mDuration > INT32_MAX) {
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return false;
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}
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for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
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if (aOther.mChannelData[channel] != AddAudioSampleOffset(mChannelData[channel],
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mBufferFormat, int32_t(mDuration))) {
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return false;
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}
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}
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}
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return true;
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}
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bool IsNull() const { return mBuffer == nullptr; }
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void SetNull(StreamTime aDuration)
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{
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mBuffer = nullptr;
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mChannelData.Clear();
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mDuration = aDuration;
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mVolume = 1.0f;
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mBufferFormat = AUDIO_FORMAT_SILENCE;
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}
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size_t ChannelCount() const { return mChannelData.Length(); }
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bool IsMuted() const { return mVolume == 0.0f; }
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size_t SizeOfExcludingThisIfUnshared(MallocSizeOf aMallocSizeOf) const
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{
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return SizeOfExcludingThis(aMallocSizeOf, true);
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}
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size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf, bool aUnshared) const
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{
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size_t amount = 0;
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// Possibly owned:
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// - mBuffer - Can hold data that is also in the decoded audio queue. If it
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// is not shared, or unshared == false it gets counted.
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if (mBuffer && (!aUnshared || !mBuffer->IsShared())) {
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amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
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}
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// Memory in the array is owned by mBuffer.
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amount += mChannelData.ShallowSizeOfExcludingThis(aMallocSizeOf);
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return amount;
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}
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template<typename T>
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const nsTArray<const T*>& ChannelData()
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{
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MOZ_ASSERT(AudioSampleTypeToFormat<T>::Format == mBufferFormat);
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return *reinterpret_cast<nsTArray<const T*>*>(&mChannelData);
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}
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StreamTime mDuration; // in frames within the buffer
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RefPtr<ThreadSharedObject> mBuffer; // the buffer object whose lifetime is managed; null means data is all zeroes
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nsTArray<const void*> mChannelData; // one pointer per channel; empty if and only if mBuffer is null
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float mVolume; // volume multiplier to apply (1.0f if mBuffer is nonnull)
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SampleFormat mBufferFormat; // format of frames in mBuffer (only meaningful if mBuffer is nonnull)
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#ifdef MOZILLA_INTERNAL_API
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mozilla::TimeStamp mTimeStamp; // time at which this has been fetched from the MediaEngine
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#endif
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};
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/**
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* A list of audio samples consisting of a sequence of slices of SharedBuffers.
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* The audio rate is determined by the track, not stored in this class.
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*/
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class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
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public:
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typedef mozilla::AudioSampleFormat SampleFormat;
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AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO) {}
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// Resample the whole segment in place.
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template<typename T>
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void Resample(SpeexResamplerState* aResampler, uint32_t aInRate, uint32_t aOutRate)
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{
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mDuration = 0;
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#ifdef DEBUG
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uint32_t segmentChannelCount = ChannelCount();
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#endif
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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nsAutoTArray<nsTArray<T>, GUESS_AUDIO_CHANNELS> output;
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nsAutoTArray<const T*, GUESS_AUDIO_CHANNELS> bufferPtrs;
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AudioChunk& c = *ci;
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// If this chunk is null, don't bother resampling, just alter its duration
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if (c.IsNull()) {
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c.mDuration = (c.mDuration * aOutRate) / aInRate;
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mDuration += c.mDuration;
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continue;
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}
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uint32_t channels = c.mChannelData.Length();
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MOZ_ASSERT(channels == segmentChannelCount);
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output.SetLength(channels);
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bufferPtrs.SetLength(channels);
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#if !defined(MOZILLA_XPCOMRT_API)
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// FIXME Bug 1126414 - XPCOMRT does not support dom::WebAudioUtils::SpeexResamplerProcess
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uint32_t inFrames = c.mDuration;
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#endif // !defined(MOZILLA_XPCOMRT_API)
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// Round up to allocate; the last frame may not be used.
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NS_ASSERTION((UINT32_MAX - aInRate + 1) / c.mDuration >= aOutRate,
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"Dropping samples");
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uint32_t outSize = (c.mDuration * aOutRate + aInRate - 1) / aInRate;
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for (uint32_t i = 0; i < channels; i++) {
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T* out = output[i].AppendElements(outSize);
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uint32_t outFrames = outSize;
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#if !defined(MOZILLA_XPCOMRT_API)
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// FIXME Bug 1126414 - XPCOMRT does not support dom::WebAudioUtils::SpeexResamplerProcess
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const T* in = static_cast<const T*>(c.mChannelData[i]);
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dom::WebAudioUtils::SpeexResamplerProcess(aResampler, i,
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in, &inFrames,
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out, &outFrames);
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MOZ_ASSERT(inFrames == c.mDuration);
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#endif // !defined(MOZILLA_XPCOMRT_API)
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bufferPtrs[i] = out;
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output[i].SetLength(outFrames);
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}
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MOZ_ASSERT(channels > 0);
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c.mDuration = output[0].Length();
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c.mBuffer = new mozilla::SharedChannelArrayBuffer<T>(&output);
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for (uint32_t i = 0; i < channels; i++) {
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c.mChannelData[i] = bufferPtrs[i];
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}
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mDuration += c.mDuration;
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}
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}
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void ResampleChunks(SpeexResamplerState* aResampler,
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uint32_t aInRate,
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uint32_t aOutRate);
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void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
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const nsTArray<const float*>& aChannelData,
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int32_t aDuration)
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{
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AudioChunk* chunk = AppendChunk(aDuration);
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chunk->mBuffer = aBuffer;
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for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
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chunk->mChannelData.AppendElement(aChannelData[channel]);
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}
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chunk->mVolume = 1.0f;
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chunk->mBufferFormat = AUDIO_FORMAT_FLOAT32;
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#ifdef MOZILLA_INTERNAL_API
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chunk->mTimeStamp = TimeStamp::Now();
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#endif
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}
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void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
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const nsTArray<const int16_t*>& aChannelData,
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int32_t aDuration)
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{
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AudioChunk* chunk = AppendChunk(aDuration);
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chunk->mBuffer = aBuffer;
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for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
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chunk->mChannelData.AppendElement(aChannelData[channel]);
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}
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chunk->mVolume = 1.0f;
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chunk->mBufferFormat = AUDIO_FORMAT_S16;
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#ifdef MOZILLA_INTERNAL_API
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chunk->mTimeStamp = TimeStamp::Now();
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#endif
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}
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// Consumes aChunk, and returns a pointer to the persistent copy of aChunk
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// in the segment.
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AudioChunk* AppendAndConsumeChunk(AudioChunk* aChunk)
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{
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AudioChunk* chunk = AppendChunk(aChunk->mDuration);
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chunk->mBuffer = aChunk->mBuffer.forget();
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chunk->mChannelData.SwapElements(aChunk->mChannelData);
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chunk->mVolume = aChunk->mVolume;
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chunk->mBufferFormat = aChunk->mBufferFormat;
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#ifdef MOZILLA_INTERNAL_API
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chunk->mTimeStamp = TimeStamp::Now();
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#endif
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return chunk;
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}
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void ApplyVolume(float aVolume);
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// Mix the segment into a mixer, interleaved. This is useful to output a
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// segment to a system audio callback. It up or down mixes to aChannelCount
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// channels.
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void WriteTo(uint64_t aID, AudioMixer& aMixer, uint32_t aChannelCount,
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uint32_t aSampleRate);
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// Mix the segment into a mixer, keeping it planar, up or down mixing to
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// aChannelCount channels.
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void Mix(AudioMixer& aMixer, uint32_t aChannelCount, uint32_t aSampleRate);
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int ChannelCount() {
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NS_WARN_IF_FALSE(!mChunks.IsEmpty(),
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"Cannot query channel count on a AudioSegment with no chunks.");
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// Find the first chunk that has non-zero channels. A chunk that hs zero
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// channels is just silence and we can simply discard it.
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for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
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if (ci->ChannelCount()) {
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return ci->ChannelCount();
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}
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}
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return 0;
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}
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bool IsNull() const {
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for (ChunkIterator ci(*const_cast<AudioSegment*>(this)); !ci.IsEnded();
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ci.Next()) {
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if (!ci->IsNull()) {
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return false;
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}
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}
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return true;
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}
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static Type StaticType() { return AUDIO; }
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virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
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{
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return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
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}
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};
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template<typename SrcT>
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void WriteChunk(AudioChunk& aChunk,
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uint32_t aOutputChannels,
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AudioDataValue* aOutputBuffer)
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{
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nsAutoTArray<const SrcT*,GUESS_AUDIO_CHANNELS> channelData;
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channelData = aChunk.ChannelData<SrcT>();
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if (channelData.Length() < aOutputChannels) {
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// Up-mix. Note that this might actually make channelData have more
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// than aOutputChannels temporarily.
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AudioChannelsUpMix(&channelData, aOutputChannels, SilentChannel::ZeroChannel<SrcT>());
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}
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if (channelData.Length() > aOutputChannels) {
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// Down-mix.
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DownmixAndInterleave(channelData, aChunk.mDuration,
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aChunk.mVolume, aOutputChannels, aOutputBuffer);
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} else {
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InterleaveAndConvertBuffer(channelData.Elements(),
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aChunk.mDuration, aChunk.mVolume,
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aOutputChannels,
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aOutputBuffer);
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}
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}
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} // namespace mozilla
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#endif /* MOZILLA_AUDIOSEGMENT_H_ */
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