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/* -*- 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_AUDIOSEGMENT_H_
#define MOZILLA_AUDIOSEGMENT_H_
#include <speex/speex_resampler.h>
#include "MediaTrackGraph.h"
#include "MediaSegment.h"
#include "AudioSampleFormat.h"
#include "AudioChannelFormat.h"
#include "SharedBuffer.h"
#include "WebAudioUtils.h"
#include "nsAutoRef.h"
#ifdef MOZILLA_INTERNAL_API
# include "mozilla/TimeStamp.h"
#endif
#include <float.h>
namespace mozilla {
struct AudioChunk;
class AudioSegment;
} // namespace mozilla
MOZ_DECLARE_RELOCATE_USING_MOVE_CONSTRUCTOR(mozilla::AudioChunk)
/**
* This allows compilation of nsTArray<AudioSegment> and
* AutoTArray<AudioSegment> since without it, static analysis fails on the
* mChunks member being a non-memmovable AutoTArray.
*
* Note that AudioSegment(const AudioSegment&) is deleted, so this should
* never come into effect.
*/
MOZ_DECLARE_RELOCATE_USING_MOVE_CONSTRUCTOR(mozilla::AudioSegment)
namespace mozilla {
template <typename T>
class SharedChannelArrayBuffer : public ThreadSharedObject {
public:
explicit SharedChannelArrayBuffer(nsTArray<nsTArray<T> >&& aBuffers)
: mBuffers(std::move(aBuffers)) {}
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
size_t amount = 0;
amount += mBuffers.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < mBuffers.Length(); i++) {
amount += mBuffers[i].ShallowSizeOfExcludingThis(aMallocSizeOf);
}
return amount;
}
size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
nsTArray<nsTArray<T> > mBuffers;
};
class AudioMixer;
/**
* For auto-arrays etc, guess this as the common number of channels.
*/
const int GUESS_AUDIO_CHANNELS = 2;
// We ensure that the graph advances in steps that are multiples of the Web
// Audio block size
const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
template <typename SrcT, typename DestT>
static void InterleaveAndConvertBuffer(const SrcT* const* aSourceChannels,
uint32_t aLength, float aVolume,
uint32_t aChannels, DestT* aOutput) {
DestT* output = aOutput;
for (size_t i = 0; i < aLength; ++i) {
for (size_t channel = 0; channel < aChannels; ++channel) {
float v = AudioSampleToFloat(aSourceChannels[channel][i]) * aVolume;
*output = FloatToAudioSample<DestT>(v);
++output;
}
}
}
template <typename SrcT, typename DestT>
static void DeinterleaveAndConvertBuffer(const SrcT* aSourceBuffer,
uint32_t aFrames, uint32_t aChannels,
DestT** aOutput) {
for (size_t i = 0; i < aChannels; i++) {
size_t interleavedIndex = i;
for (size_t j = 0; j < aFrames; j++) {
ConvertAudioSample(aSourceBuffer[interleavedIndex], aOutput[i][j]);
interleavedIndex += aChannels;
}
}
}
class SilentChannel {
public:
static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
static const uint8_t
gZeroChannel[MAX_AUDIO_SAMPLE_SIZE * AUDIO_PROCESSING_FRAMES];
// We take advantage of the fact that zero in float and zero in int have the
// same all-zeros bit layout.
template <typename T>
static const T* ZeroChannel();
};
/**
* Given an array of input channels (aChannelData), downmix to aOutputChannels,
* interleave the channel data. A total of aOutputChannels*aDuration
* interleaved samples will be copied to a channel buffer in aOutput.
*/
template <typename SrcT, typename DestT>
void DownmixAndInterleave(const nsTArray<const SrcT*>& aChannelData,
int32_t aDuration, float aVolume,
uint32_t aOutputChannels, DestT* aOutput) {
if (aChannelData.Length() == aOutputChannels) {
InterleaveAndConvertBuffer(aChannelData.Elements(), aDuration, aVolume,
aOutputChannels, aOutput);
} else {
AutoTArray<SrcT*, GUESS_AUDIO_CHANNELS> outputChannelData;
AutoTArray<SrcT,
SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
outputBuffers;
outputChannelData.SetLength(aOutputChannels);
outputBuffers.SetLength(aDuration * aOutputChannels);
for (uint32_t i = 0; i < aOutputChannels; i++) {
outputChannelData[i] = outputBuffers.Elements() + aDuration * i;
}
AudioChannelsDownMix(aChannelData, outputChannelData.Elements(),
aOutputChannels, aDuration);
InterleaveAndConvertBuffer(outputChannelData.Elements(), aDuration, aVolume,
aOutputChannels, aOutput);
}
}
/**
* An AudioChunk represents a multi-channel buffer of audio samples.
* It references an underlying ThreadSharedObject which manages the lifetime
* of the buffer. An AudioChunk maintains its own duration and channel data
* pointers so it can represent a subinterval of a buffer without copying.
* An AudioChunk can store its individual channels anywhere; it maintains
* separate pointers to each channel's buffer.
*/
struct AudioChunk {
typedef mozilla::AudioSampleFormat SampleFormat;
// Generic methods
void SliceTo(TrackTime aStart, TrackTime aEnd) {
MOZ_ASSERT(aStart >= 0 && aStart < aEnd && aEnd <= mDuration,
"Slice out of bounds");
if (mBuffer) {
MOZ_ASSERT(aStart < INT32_MAX,
"Can't slice beyond 32-bit sample lengths");
for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
mChannelData[channel] = AddAudioSampleOffset(
mChannelData[channel], mBufferFormat, int32_t(aStart));
}
}
mDuration = aEnd - aStart;
}
TrackTime GetDuration() const { return mDuration; }
bool CanCombineWithFollowing(const AudioChunk& aOther) const {
if (aOther.mBuffer != mBuffer) {
return false;
}
if (!mBuffer) {
return true;
}
if (aOther.mVolume != mVolume) {
return false;
}
if (aOther.mPrincipalHandle != mPrincipalHandle) {
return false;
}
NS_ASSERTION(aOther.mBufferFormat == mBufferFormat,
"Wrong metadata about buffer");
NS_ASSERTION(aOther.mChannelData.Length() == mChannelData.Length(),
"Mismatched channel count");
if (mDuration > INT32_MAX) {
return false;
}
for (uint32_t channel = 0; channel < mChannelData.Length(); ++channel) {
if (aOther.mChannelData[channel] !=
AddAudioSampleOffset(mChannelData[channel], mBufferFormat,
int32_t(mDuration))) {
return false;
}
}
return true;
}
bool IsNull() const { return mBuffer == nullptr; }
void SetNull(TrackTime aDuration) {
mBuffer = nullptr;
mChannelData.Clear();
mDuration = aDuration;
mVolume = 1.0f;
mBufferFormat = AUDIO_FORMAT_SILENCE;
mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
}
uint32_t ChannelCount() const { return mChannelData.Length(); }
bool IsMuted() const { return mVolume == 0.0f; }
size_t SizeOfExcludingThisIfUnshared(MallocSizeOf aMallocSizeOf) const {
return SizeOfExcludingThis(aMallocSizeOf, true);
}
size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf, bool aUnshared) const {
size_t amount = 0;
// Possibly owned:
// - mBuffer - Can hold data that is also in the decoded audio queue. If it
// is not shared, or unshared == false it gets counted.
if (mBuffer && (!aUnshared || !mBuffer->IsShared())) {
amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
}
// Memory in the array is owned by mBuffer.
amount += mChannelData.ShallowSizeOfExcludingThis(aMallocSizeOf);
return amount;
}
template <typename T>
const nsTArray<const T*>& ChannelData() const {
MOZ_ASSERT(AudioSampleTypeToFormat<T>::Format == mBufferFormat);
return *reinterpret_cast<const AutoTArray<const T*, GUESS_AUDIO_CHANNELS>*>(
&mChannelData);
}
/**
* ChannelFloatsForWrite() should be used only when mBuffer is owned solely
* by the calling thread.
*/
template <typename T>
T* ChannelDataForWrite(size_t aChannel) {
MOZ_ASSERT(AudioSampleTypeToFormat<T>::Format == mBufferFormat);
MOZ_ASSERT(!mBuffer->IsShared());
return static_cast<T*>(const_cast<void*>(mChannelData[aChannel]));
}
const PrincipalHandle& GetPrincipalHandle() const { return mPrincipalHandle; }
TrackTime mDuration = 0; // in frames within the buffer
RefPtr<ThreadSharedObject> mBuffer; // the buffer object whose lifetime is
// managed; null means data is all zeroes
// one pointer per channel; empty if and only if mBuffer is null
CopyableAutoTArray<const void*, GUESS_AUDIO_CHANNELS> mChannelData;
float mVolume = 1.0f; // volume multiplier to apply
// format of frames in mBuffer (or silence if mBuffer is null)
SampleFormat mBufferFormat = AUDIO_FORMAT_SILENCE;
// principalHandle for the data in this chunk.
// This can be compared to an nsIPrincipal* when back on main thread.
PrincipalHandle mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
};
/**
* A list of audio samples consisting of a sequence of slices of SharedBuffers.
* The audio rate is determined by the track, not stored in this class.
*/
class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
public:
typedef mozilla::AudioSampleFormat SampleFormat;
AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO) {}
AudioSegment(AudioSegment&& aSegment) = default;
AudioSegment(const AudioSegment&) = delete;
AudioSegment& operator=(const AudioSegment&) = delete;
~AudioSegment() = default;
// Resample the whole segment in place. `aResampler` is an instance of a
// resampler, initialized with `aResamplerChannelCount` channels. If this
// function finds a chunk with more channels, `aResampler` is destroyed and a
// new resampler is created, and `aResamplerChannelCount` is updated with the
// new channel count value.
template <typename T>
void Resample(nsAutoRef<SpeexResamplerState>& aResampler,
uint32_t* aResamplerChannelCount, uint32_t aInRate,
uint32_t aOutRate) {
mDuration = 0;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AutoTArray<nsTArray<T>, GUESS_AUDIO_CHANNELS> output;
AutoTArray<const T*, GUESS_AUDIO_CHANNELS> bufferPtrs;
AudioChunk& c = *ci;
// If this chunk is null, don't bother resampling, just alter its duration
if (c.IsNull()) {
c.mDuration = (c.mDuration * aOutRate) / aInRate;
mDuration += c.mDuration;
continue;
}
uint32_t channels = c.mChannelData.Length();
// This might introduce a discontinuity, but a channel count change in the
// middle of a stream is not that common. This also initializes the
// resampler as late as possible.
if (channels != *aResamplerChannelCount) {
SpeexResamplerState* state =
speex_resampler_init(channels, aInRate, aOutRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
MOZ_ASSERT(state);
aResampler.own(state);
*aResamplerChannelCount = channels;
}
output.SetLength(channels);
bufferPtrs.SetLength(channels);
uint32_t inFrames = c.mDuration;
// Round up to allocate; the last frame may not be used.
NS_ASSERTION((UINT32_MAX - aInRate + 1) / c.mDuration >= aOutRate,
"Dropping samples");
uint32_t outSize = (c.mDuration * aOutRate + aInRate - 1) / aInRate;
for (uint32_t i = 0; i < channels; i++) {
T* out = output[i].AppendElements(outSize);
uint32_t outFrames = outSize;
const T* in = static_cast<const T*>(c.mChannelData[i]);
dom::WebAudioUtils::SpeexResamplerProcess(aResampler.get(), i, in,
&inFrames, out, &outFrames);
MOZ_ASSERT(inFrames == c.mDuration);
bufferPtrs[i] = out;
output[i].SetLength(outFrames);
}
MOZ_ASSERT(channels > 0);
c.mDuration = output[0].Length();
c.mBuffer = new mozilla::SharedChannelArrayBuffer<T>(std::move(output));
for (uint32_t i = 0; i < channels; i++) {
c.mChannelData[i] = bufferPtrs[i];
}
mDuration += c.mDuration;
}
}
void ResampleChunks(nsAutoRef<SpeexResamplerState>& aResampler,
uint32_t* aResamplerChannelCount, uint32_t aInRate,
uint32_t aOutRate);
void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
const nsTArray<const float*>& aChannelData,
int32_t aDuration,
const PrincipalHandle& aPrincipalHandle) {
AudioChunk* chunk = AppendChunk(aDuration);
chunk->mBuffer = aBuffer;
MOZ_ASSERT(chunk->mBuffer || aChannelData.IsEmpty(),
"Appending invalid data ?");
for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
chunk->mChannelData.AppendElement(aChannelData[channel]);
}
chunk->mBufferFormat = AUDIO_FORMAT_FLOAT32;
chunk->mPrincipalHandle = aPrincipalHandle;
}
void AppendFrames(already_AddRefed<ThreadSharedObject> aBuffer,
const nsTArray<const int16_t*>& aChannelData,
int32_t aDuration,
const PrincipalHandle& aPrincipalHandle) {
AudioChunk* chunk = AppendChunk(aDuration);
chunk->mBuffer = aBuffer;
MOZ_ASSERT(chunk->mBuffer || aChannelData.IsEmpty(),
"Appending invalid data ?");
for (uint32_t channel = 0; channel < aChannelData.Length(); ++channel) {
chunk->mChannelData.AppendElement(aChannelData[channel]);
}
chunk->mBufferFormat = AUDIO_FORMAT_S16;
chunk->mPrincipalHandle = aPrincipalHandle;
}
// Consumes aChunk, and returns a pointer to the persistent copy of aChunk
// in the segment.
AudioChunk* AppendAndConsumeChunk(AudioChunk* aChunk) {
AudioChunk* chunk = AppendChunk(aChunk->mDuration);
chunk->mBuffer = std::move(aChunk->mBuffer);
chunk->mChannelData = std::move(aChunk->mChannelData);
MOZ_ASSERT(chunk->mBuffer || aChunk->mChannelData.IsEmpty(),
"Appending invalid data ?");
chunk->mVolume = aChunk->mVolume;
chunk->mBufferFormat = aChunk->mBufferFormat;
chunk->mPrincipalHandle = aChunk->mPrincipalHandle;
return chunk;
}
void ApplyVolume(float aVolume);
// Mix the segment into a mixer, interleaved. This is useful to output a
// segment to a system audio callback. It up or down mixes to aChannelCount
// channels.
void WriteTo(AudioMixer& aMixer, uint32_t aChannelCount,
uint32_t aSampleRate);
// Mix the segment into a mixer, keeping it planar, up or down mixing to
// aChannelCount channels.
void Mix(AudioMixer& aMixer, uint32_t aChannelCount, uint32_t aSampleRate);
// Returns the maximum
uint32_t MaxChannelCount() {
// Find the first chunk that has non-zero channels. A chunk that hs zero
// channels is just silence and we can simply discard it.
uint32_t channelCount = 0;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
if (ci->ChannelCount()) {
channelCount = std::max(channelCount, ci->ChannelCount());
}
}
return channelCount;
}
static Type StaticType() { return AUDIO; }
size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
};
template <typename SrcT>
void WriteChunk(AudioChunk& aChunk, uint32_t aOutputChannels,
AudioDataValue* aOutputBuffer) {
AutoTArray<const SrcT*, GUESS_AUDIO_CHANNELS> channelData;
channelData = aChunk.ChannelData<SrcT>().Clone();
if (channelData.Length() < aOutputChannels) {
// Up-mix. Note that this might actually make channelData have more
// than aOutputChannels temporarily.
AudioChannelsUpMix(&channelData, aOutputChannels,
SilentChannel::ZeroChannel<SrcT>());
}
if (channelData.Length() > aOutputChannels) {
// Down-mix.
DownmixAndInterleave(channelData, aChunk.mDuration, aChunk.mVolume,
aOutputChannels, aOutputBuffer);
} else {
InterleaveAndConvertBuffer(channelData.Elements(), aChunk.mDuration,
aChunk.mVolume, aOutputChannels, aOutputBuffer);
}
}
} // namespace mozilla
#endif /* MOZILLA_AUDIOSEGMENT_H_ */