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author | Matt A. Tobin <mattatobin@localhost.localdomain> | 2018-02-02 04:16:08 -0500 |
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committer | Matt A. Tobin <mattatobin@localhost.localdomain> | 2018-02-02 04:16:08 -0500 |
commit | 5f8de423f190bbb79a62f804151bc24824fa32d8 (patch) | |
tree | 10027f336435511475e392454359edea8e25895d /dom/media/webaudio/blink/ReverbConvolver.cpp | |
parent | 49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff) | |
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Add m-esr52 at 52.6.0
Diffstat (limited to 'dom/media/webaudio/blink/ReverbConvolver.cpp')
-rw-r--r-- | dom/media/webaudio/blink/ReverbConvolver.cpp | 265 |
1 files changed, 265 insertions, 0 deletions
diff --git a/dom/media/webaudio/blink/ReverbConvolver.cpp b/dom/media/webaudio/blink/ReverbConvolver.cpp new file mode 100644 index 000000000..e739400ae --- /dev/null +++ b/dom/media/webaudio/blink/ReverbConvolver.cpp @@ -0,0 +1,265 @@ +/* + * 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) + : m_impulseResponseLength(impulseResponseLength) + , m_accumulationBuffer(impulseResponseLength + WEBAUDIO_BLOCK_SIZE) + , m_inputBuffer(InputBufferSize) + , m_backgroundThread("ConvolverWorker") + , m_backgroundThreadCondition(&m_backgroundThreadLock) + , m_useBackgroundThreads(useBackgroundThreads) + , m_wantsToExit(false) + , m_moreInputBuffered(false) +{ + // 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; + + nsAutoPtr<ReverbConvolverStage> stage + (new ReverbConvolverStage(response, totalResponseLength, + reverbTotalLatency, stageOffset, stageSize, + fftSize, renderPhase, + &m_accumulationBuffer)); + + bool isBackgroundStage = false; + + if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) { + m_backgroundStages.AppendElement(stage.forget()); + isBackgroundStage = true; + } else + m_stages.AppendElement(stage.forget()); + + // 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(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 + { + AutoLock locker(m_backgroundThreadLock); + m_moreInputBuffered = true; + m_backgroundThreadCondition.Signal(); + } + + 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_backgroundThreadLock + // - m_backgroundThreadCondition + return amount; +} + +void ReverbConvolver::backgroundThreadEntry() +{ + while (!m_wantsToExit) { + // Wait for realtime thread to give us more input + m_moreInputBuffered = false; + { + AutoLock locker(m_backgroundThreadLock); + while (!m_moreInputBuffered && !m_wantsToExit) + m_backgroundThreadCondition.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 MutexLocker 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_backgroundThreadLock.Try()) { + m_moreInputBuffered = true; + m_backgroundThreadCondition.Signal(); + m_backgroundThreadLock.Release(); + } +} + +} // namespace WebCore |