gecko-dev/dom/media/webaudio/DelayBuffer.cpp

236 строки
8.5 KiB
C++

/* -*- 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/. */
#include "DelayBuffer.h"
#include "mozilla/PodOperations.h"
#include "AudioChannelFormat.h"
#include "AudioNodeEngine.h"
namespace mozilla {
size_t DelayBuffer::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
size_t amount = 0;
amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < mChunks.Length(); i++) {
amount += mChunks[i].SizeOfExcludingThis(aMallocSizeOf, false);
}
amount += mUpmixChannels.ShallowSizeOfExcludingThis(aMallocSizeOf);
return amount;
}
void DelayBuffer::Write(const AudioBlock& aInputChunk) {
// We must have a reference to the buffer if there are channels
MOZ_ASSERT(aInputChunk.IsNull() == !aInputChunk.ChannelCount());
#ifdef DEBUG
MOZ_ASSERT(!mHaveWrittenBlock);
mHaveWrittenBlock = true;
#endif
if (!EnsureBuffer()) {
return;
}
if (mCurrentChunk == mLastReadChunk) {
mLastReadChunk = -1; // invalidate cache
}
mChunks[mCurrentChunk] = aInputChunk.AsAudioChunk();
}
void DelayBuffer::Read(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk,
ChannelInterpretation aChannelInterpretation) {
int chunkCount = mChunks.Length();
if (!chunkCount) {
aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
// Find the maximum number of contributing channels to determine the output
// channel count that retains all signal information. Buffered blocks will
// be upmixed if necessary.
//
// First find the range of "delay" offsets backwards from the current
// position. Note that these may be negative for frames that are after the
// current position (including i).
float minDelay = aPerFrameDelays[0];
float maxDelay = minDelay;
for (unsigned i = 1; i < WEBAUDIO_BLOCK_SIZE; ++i) {
minDelay = std::min(minDelay, aPerFrameDelays[i] - i);
maxDelay = std::max(maxDelay, aPerFrameDelays[i] - i);
}
// Now find the chunks touched by this range and check their channel counts.
int oldestChunk = ChunkForDelay(std::ceil(maxDelay));
int youngestChunk = ChunkForDelay(std::floor(minDelay));
uint32_t channelCount = 0;
for (int i = oldestChunk; true; i = (i + 1) % chunkCount) {
channelCount =
GetAudioChannelsSuperset(channelCount, mChunks[i].ChannelCount());
if (i == youngestChunk) {
break;
}
}
if (channelCount) {
aOutputChunk->AllocateChannels(channelCount);
ReadChannels(aPerFrameDelays, aOutputChunk, 0, channelCount,
aChannelInterpretation);
} else {
aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
}
}
void DelayBuffer::ReadChannel(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk, uint32_t aChannel,
ChannelInterpretation aChannelInterpretation) {
if (!mChunks.Length()) {
float* outputChannel = aOutputChunk->ChannelFloatsForWrite(aChannel);
PodZero(outputChannel, WEBAUDIO_BLOCK_SIZE);
return;
}
ReadChannels(aPerFrameDelays, aOutputChunk, aChannel, 1,
aChannelInterpretation);
}
void DelayBuffer::ReadChannels(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioBlock* aOutputChunk, uint32_t aFirstChannel,
uint32_t aNumChannelsToRead,
ChannelInterpretation aChannelInterpretation) {
uint32_t totalChannelCount = aOutputChunk->ChannelCount();
uint32_t readChannelsEnd = aFirstChannel + aNumChannelsToRead;
MOZ_ASSERT(readChannelsEnd <= totalChannelCount);
if (mUpmixChannels.Length() != totalChannelCount) {
mLastReadChunk = -1; // invalidate cache
}
for (uint32_t channel = aFirstChannel; channel < readChannelsEnd; ++channel) {
PodZero(aOutputChunk->ChannelFloatsForWrite(channel), WEBAUDIO_BLOCK_SIZE);
}
for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
float currentDelay = aPerFrameDelays[i];
MOZ_ASSERT(currentDelay >= 0.0f);
MOZ_ASSERT(currentDelay <= (mChunks.Length() - 1) * WEBAUDIO_BLOCK_SIZE);
// Interpolate two input frames in case the read position does not match
// an integer index.
// Use the larger delay, for the older frame, first, as this is more
// likely to use the cached upmixed channel arrays.
int floorDelay = int(currentDelay);
float interpolationFactor = currentDelay - floorDelay;
int positions[2];
positions[1] = PositionForDelay(floorDelay) + i;
positions[0] = positions[1] - 1;
for (unsigned tick = 0; tick < ArrayLength(positions); ++tick) {
int readChunk = ChunkForPosition(positions[tick]);
// The zero check on interpolationFactor is important because, when
// currentDelay is integer, positions[0] may be outside the range
// considered for determining totalChannelCount.
// mVolume is not set on default initialized chunks so also handle null
// chunks specially.
if (interpolationFactor != 0.0f && !mChunks[readChunk].IsNull()) {
int readOffset = OffsetForPosition(positions[tick]);
UpdateUpmixChannels(readChunk, totalChannelCount,
aChannelInterpretation);
float multiplier = interpolationFactor * mChunks[readChunk].mVolume;
for (uint32_t channel = aFirstChannel; channel < readChannelsEnd;
++channel) {
aOutputChunk->ChannelFloatsForWrite(channel)[i] +=
multiplier * mUpmixChannels[channel][readOffset];
}
}
interpolationFactor = 1.0f - interpolationFactor;
}
}
}
void DelayBuffer::Read(float aDelayTicks, AudioBlock* aOutputChunk,
ChannelInterpretation aChannelInterpretation) {
float computedDelay[WEBAUDIO_BLOCK_SIZE];
for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
computedDelay[i] = aDelayTicks;
}
Read(computedDelay, aOutputChunk, aChannelInterpretation);
}
bool DelayBuffer::EnsureBuffer() {
if (mChunks.Length() == 0) {
// The length of the buffer is at least one block greater than the maximum
// delay so that writing an input block does not overwrite the block that
// would subsequently be read at maximum delay. Also round up to the next
// block size, so that no block of writes will need to wrap.
const int chunkCount = (mMaxDelayTicks + 2 * WEBAUDIO_BLOCK_SIZE - 1) >>
WEBAUDIO_BLOCK_SIZE_BITS;
if (!mChunks.SetLength(chunkCount, fallible)) {
return false;
}
mLastReadChunk = -1;
}
return true;
}
int DelayBuffer::PositionForDelay(int aDelay) {
// Adding mChunks.Length() keeps integers positive for defined and
// appropriate bitshift, remainder, and bitwise operations.
return ((mCurrentChunk + mChunks.Length()) * WEBAUDIO_BLOCK_SIZE) - aDelay;
}
int DelayBuffer::ChunkForPosition(int aPosition) {
MOZ_ASSERT(aPosition >= 0);
return (aPosition >> WEBAUDIO_BLOCK_SIZE_BITS) % mChunks.Length();
}
int DelayBuffer::OffsetForPosition(int aPosition) {
MOZ_ASSERT(aPosition >= 0);
return aPosition & (WEBAUDIO_BLOCK_SIZE - 1);
}
int DelayBuffer::ChunkForDelay(int aDelay) {
return ChunkForPosition(PositionForDelay(aDelay));
}
void DelayBuffer::UpdateUpmixChannels(
int aNewReadChunk, uint32_t aChannelCount,
ChannelInterpretation aChannelInterpretation) {
if (aNewReadChunk == mLastReadChunk) {
MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount);
return;
}
NS_WARNING_ASSERTION(mHaveWrittenBlock || aNewReadChunk != mCurrentChunk,
"Smoothing is making feedback delay too small.");
mLastReadChunk = aNewReadChunk;
mUpmixChannels = mChunks[aNewReadChunk].ChannelData<float>();
MOZ_ASSERT(mUpmixChannels.Length() <= aChannelCount);
if (mUpmixChannels.Length() < aChannelCount) {
if (aChannelInterpretation == ChannelInterpretation::Speakers) {
AudioChannelsUpMix(&mUpmixChannels, aChannelCount,
SilentChannel::ZeroChannel<float>());
MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount,
"We called GetAudioChannelsSuperset to avoid this");
} else {
// Fill up the remaining channels with zeros
for (uint32_t channel = mUpmixChannels.Length(); channel < aChannelCount;
++channel) {
mUpmixChannels.AppendElement(SilentChannel::ZeroChannel<float>());
}
}
}
}
} // namespace mozilla