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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /dom/media/webaudio/DelayBuffer.cpp | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'dom/media/webaudio/DelayBuffer.cpp')
-rw-r--r-- | dom/media/webaudio/DelayBuffer.cpp | 236 |
1 files changed, 236 insertions, 0 deletions
diff --git a/dom/media/webaudio/DelayBuffer.cpp b/dom/media/webaudio/DelayBuffer.cpp new file mode 100644 index 0000000000..52c8d4184b --- /dev/null +++ b/dom/media/webaudio/DelayBuffer.cpp @@ -0,0 +1,236 @@ +/* -*- 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.ClearAndRetainStorage(); + mUpmixChannels.AppendElements(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 |