From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- dom/media/webaudio/blink/ReverbConvolver.cpp | 273 +++++++++++++++++++++++++++ 1 file changed, 273 insertions(+) create mode 100644 dom/media/webaudio/blink/ReverbConvolver.cpp (limited to 'dom/media/webaudio/blink/ReverbConvolver.cpp') diff --git a/dom/media/webaudio/blink/ReverbConvolver.cpp b/dom/media/webaudio/blink/ReverbConvolver.cpp new file mode 100644 index 0000000000..6965a28535 --- /dev/null +++ b/dom/media/webaudio/blink/ReverbConvolver.cpp @@ -0,0 +1,273 @@ +/* + * Copyright (C) 2010 Google Inc. All rights reserved. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of + * its contributors may be used to endorse or promote products derived + * from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY + * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED + * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY + * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; + * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND + * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT + * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF + * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + */ + +#include "ReverbConvolver.h" +#include "ReverbConvolverStage.h" + +using namespace mozilla; + +namespace WebCore { + +const int InputBufferSize = 8 * 16384; + +// We only process the leading portion of the impulse response in the real-time +// thread. We don't exceed this length. It turns out then, that the background +// thread has about 278msec of scheduling slop. Empirically, this has been found +// to be a good compromise between giving enough time for scheduling slop, while +// still minimizing the amount of processing done in the primary (high-priority) +// thread. This was found to be a good value on Mac OS X, and may work well on +// other platforms as well, assuming the very rough scheduling latencies are +// similar on these time-scales. Of course, this code may need to be tuned for +// individual platforms if this assumption is found to be incorrect. +const size_t RealtimeFrameLimit = 8192 + 4096 // ~278msec @ 44.1KHz + - WEBAUDIO_BLOCK_SIZE; +// First stage will have size MinFFTSize - successive stages will double in +// size each time until we hit the maximum size. +const size_t MinFFTSize = 256; +// If we are using background threads then don't exceed this FFT size for the +// stages which run in the real-time thread. This avoids having only one or +// two large stages (size 16384 or so) at the end which take a lot of time +// every several processing slices. This way we amortize the cost over more +// processing slices. +const size_t MaxRealtimeFFTSize = 4096; + +ReverbConvolver::ReverbConvolver(const float* impulseResponseData, + size_t impulseResponseLength, + size_t maxFFTSize, size_t convolverRenderPhase, + bool useBackgroundThreads, + bool* aAllocationFailure) + : m_impulseResponseLength(impulseResponseLength), + m_accumulationBuffer(), + m_inputBuffer(InputBufferSize), + m_backgroundThread("ConvolverWorker"), + m_backgroundThreadMonitor("ConvolverMonitor"), + m_useBackgroundThreads(useBackgroundThreads), + m_wantsToExit(false), + m_moreInputBuffered(false) { + *aAllocationFailure = !m_accumulationBuffer.allocate(impulseResponseLength + + WEBAUDIO_BLOCK_SIZE); + if (*aAllocationFailure) { + return; + } + // For the moment, a good way to know if we have real-time constraint is to + // check if we're using background threads. Otherwise, assume we're being run + // from a command-line tool. + bool hasRealtimeConstraint = useBackgroundThreads; + + const float* response = impulseResponseData; + size_t totalResponseLength = impulseResponseLength; + + // The total latency is zero because the first FFT stage is small enough + // to return output in the first block. + size_t reverbTotalLatency = 0; + + size_t stageOffset = 0; + size_t stagePhase = 0; + size_t fftSize = MinFFTSize; + while (stageOffset < totalResponseLength) { + size_t stageSize = fftSize / 2; + + // For the last stage, it's possible that stageOffset is such that we're + // straddling the end of the impulse response buffer (if we use stageSize), + // so reduce the last stage's length... + if (stageSize + stageOffset > totalResponseLength) { + stageSize = totalResponseLength - stageOffset; + // Use smallest FFT that is large enough to cover the last stage. + fftSize = MinFFTSize; + while (stageSize * 2 > fftSize) { + fftSize *= 2; + } + } + + // This "staggers" the time when each FFT happens so they don't all happen + // at the same time + int renderPhase = convolverRenderPhase + stagePhase; + + UniquePtr stage(new ReverbConvolverStage( + response, totalResponseLength, reverbTotalLatency, stageOffset, + stageSize, fftSize, renderPhase, &m_accumulationBuffer)); + + bool isBackgroundStage = false; + + if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) { + m_backgroundStages.AppendElement(std::move(stage)); + isBackgroundStage = true; + } else + m_stages.AppendElement(std::move(stage)); + + // Figure out next FFT size + fftSize *= 2; + + stageOffset += stageSize; + + if (hasRealtimeConstraint && !isBackgroundStage && + fftSize > MaxRealtimeFFTSize) { + fftSize = MaxRealtimeFFTSize; + // Custom phase positions for all but the first of the realtime + // stages of largest size. These spread out the work of the + // larger realtime stages. None of the FFTs of size 1024, 2048 or + // 4096 are performed when processing the same block. The first + // MaxRealtimeFFTSize = 4096 stage, at the end of the doubling, + // performs its FFT at block 7. The FFTs of size 2048 are + // performed in blocks 3 + 8 * n and size 1024 at 1 + 4 * n. + const uint32_t phaseLookup[] = {14, 0, 10, 4}; + stagePhase = WEBAUDIO_BLOCK_SIZE * + phaseLookup[m_stages.Length() % ArrayLength(phaseLookup)]; + } else if (fftSize > maxFFTSize) { + fftSize = maxFFTSize; + // A prime offset spreads out FFTs in a way that all + // available phase positions will be used if there are sufficient + // stages. + stagePhase += 5 * WEBAUDIO_BLOCK_SIZE; + } else if (stageSize > WEBAUDIO_BLOCK_SIZE) { + // As the stages are doubling in size, the next FFT will occur + // mid-way between FFTs for this stage. + stagePhase = stageSize - WEBAUDIO_BLOCK_SIZE; + } + } + + // Start up background thread + // FIXME: would be better to up the thread priority here. It doesn't need to + // be real-time, but higher than the default... + if (this->useBackgroundThreads() && m_backgroundStages.Length() > 0) { + if (!m_backgroundThread.Start()) { + NS_WARNING("Cannot start convolver thread."); + return; + } + m_backgroundThread.message_loop()->PostTask(NewNonOwningRunnableMethod( + "WebCore::ReverbConvolver::backgroundThreadEntry", this, + &ReverbConvolver::backgroundThreadEntry)); + } +} + +ReverbConvolver::~ReverbConvolver() { + // Wait for background thread to stop + if (useBackgroundThreads() && m_backgroundThread.IsRunning()) { + m_wantsToExit = true; + + // Wake up thread so it can return + { + MonitorAutoLock locker(m_backgroundThreadMonitor); + m_moreInputBuffered = true; + m_backgroundThreadMonitor.Notify(); + } + + m_backgroundThread.Stop(); + } +} + +size_t ReverbConvolver::sizeOfIncludingThis( + mozilla::MallocSizeOf aMallocSizeOf) const { + size_t amount = aMallocSizeOf(this); + amount += m_stages.ShallowSizeOfExcludingThis(aMallocSizeOf); + for (size_t i = 0; i < m_stages.Length(); i++) { + if (m_stages[i]) { + amount += m_stages[i]->sizeOfIncludingThis(aMallocSizeOf); + } + } + + amount += m_backgroundStages.ShallowSizeOfExcludingThis(aMallocSizeOf); + for (size_t i = 0; i < m_backgroundStages.Length(); i++) { + if (m_backgroundStages[i]) { + amount += m_backgroundStages[i]->sizeOfIncludingThis(aMallocSizeOf); + } + } + + // NB: The buffer sizes are static, so even though they might be accessed + // in another thread it's safe to measure them. + amount += m_accumulationBuffer.sizeOfExcludingThis(aMallocSizeOf); + amount += m_inputBuffer.sizeOfExcludingThis(aMallocSizeOf); + + // Possible future measurements: + // - m_backgroundThread + // - m_backgroundThreadMonitor + return amount; +} + +void ReverbConvolver::backgroundThreadEntry() { + while (!m_wantsToExit) { + // Wait for realtime thread to give us more input + m_moreInputBuffered = false; + { + MonitorAutoLock locker(m_backgroundThreadMonitor); + while (!m_moreInputBuffered && !m_wantsToExit) + m_backgroundThreadMonitor.Wait(); + } + + // Process all of the stages until their read indices reach the input + // buffer's write index + int writeIndex = m_inputBuffer.writeIndex(); + + // Even though it doesn't seem like every stage needs to maintain its own + // version of readIndex we do this in case we want to run in more than one + // background thread. + int readIndex; + + while ((readIndex = m_backgroundStages[0]->inputReadIndex()) != + writeIndex) { // FIXME: do better to detect buffer overrun... + // Accumulate contributions from each stage + for (size_t i = 0; i < m_backgroundStages.Length(); ++i) + m_backgroundStages[i]->processInBackground(this); + } + } +} + +void ReverbConvolver::process(const float* sourceChannelData, + float* destinationChannelData) { + const float* source = sourceChannelData; + float* destination = destinationChannelData; + bool isDataSafe = source && destination; + MOZ_ASSERT(isDataSafe); + if (!isDataSafe) return; + + // Feed input buffer (read by all threads) + m_inputBuffer.write(source, WEBAUDIO_BLOCK_SIZE); + + // Accumulate contributions from each stage + for (size_t i = 0; i < m_stages.Length(); ++i) m_stages[i]->process(source); + + // Finally read from accumulation buffer + m_accumulationBuffer.readAndClear(destination, WEBAUDIO_BLOCK_SIZE); + + // Now that we've buffered more input, wake up our background thread. + + // Not using a MonitorAutoLock looks strange, but we use a TryLock() instead + // because this is run on the real-time thread where it is a disaster for the + // lock to be contended (causes audio glitching). It's OK if we fail to + // signal from time to time, since we'll get to it the next time we're called. + // We're called repeatedly and frequently (around every 3ms). The background + // thread is processing well into the future and has a considerable amount of + // leeway here... + if (m_backgroundThreadMonitor.TryLock()) { + m_moreInputBuffered = true; + m_backgroundThreadMonitor.Notify(); + m_backgroundThreadMonitor.Unlock(); + } +} + +} // namespace WebCore -- cgit v1.2.3