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author | Steve Block <steveblock@google.com> | 2011-05-06 11:45:16 +0100 |
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committer | Steve Block <steveblock@google.com> | 2011-05-12 13:44:10 +0100 |
commit | cad810f21b803229eb11403f9209855525a25d57 (patch) | |
tree | 29a6fd0279be608e0fe9ffe9841f722f0f4e4269 /Source/WebCore/platform/audio/ReverbConvolver.cpp | |
parent | 121b0cf4517156d0ac5111caf9830c51b69bae8f (diff) | |
download | external_webkit-cad810f21b803229eb11403f9209855525a25d57.zip external_webkit-cad810f21b803229eb11403f9209855525a25d57.tar.gz external_webkit-cad810f21b803229eb11403f9209855525a25d57.tar.bz2 |
Merge WebKit at r75315: Initial merge by git.
Change-Id: I570314b346ce101c935ed22a626b48c2af266b84
Diffstat (limited to 'Source/WebCore/platform/audio/ReverbConvolver.cpp')
-rw-r--r-- | Source/WebCore/platform/audio/ReverbConvolver.cpp | 230 |
1 files changed, 230 insertions, 0 deletions
diff --git a/Source/WebCore/platform/audio/ReverbConvolver.cpp b/Source/WebCore/platform/audio/ReverbConvolver.cpp new file mode 100644 index 0000000..120c9f2 --- /dev/null +++ b/Source/WebCore/platform/audio/ReverbConvolver.cpp @@ -0,0 +1,230 @@ +/* + * 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 "config.h" + +#if ENABLE(WEB_AUDIO) + +#include "ReverbConvolver.h" + +#include "VectorMath.h" +#include "AudioBus.h" + +namespace WebCore { + +using namespace VectorMath; + +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 + +const size_t MinFFTSize = 256; +const size_t MaxRealtimeFFTSize = 2048; + +static void* backgroundThreadEntry(void* threadData) +{ + ReverbConvolver* reverbConvolver = static_cast<ReverbConvolver*>(threadData); + reverbConvolver->backgroundThreadEntry(); + return 0; +} + +ReverbConvolver::ReverbConvolver(AudioChannel* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t convolverRenderPhase, bool useBackgroundThreads) + : m_impulseResponseLength(impulseResponse->length()) + , m_accumulationBuffer(impulseResponse->length() + renderSliceSize) + , m_inputBuffer(InputBufferSize) + , m_renderSliceSize(renderSliceSize) + , m_minFFTSize(MinFFTSize) // First stage will have this size - successive stages will double in size each time + , m_maxFFTSize(maxFFTSize) // until we hit m_maxFFTSize + , m_useBackgroundThreads(useBackgroundThreads) + , m_backgroundThread(0) + , m_wantsToExit(false) + , m_moreInputBuffered(false) +{ + // 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. + m_maxRealtimeFFTSize = MaxRealtimeFFTSize; + + // 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; + + float* response = impulseResponse->data(); + size_t totalResponseLength = impulseResponse->length(); + + // Because we're not using direct-convolution in the leading portion, the reverb has an overall latency of half the first-stage FFT size + size_t reverbTotalLatency = m_minFFTSize / 2; + + size_t stageOffset = 0; + int i = 0; + size_t fftSize = m_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; + + // This "staggers" the time when each FFT happens so they don't all happen at the same time + int renderPhase = convolverRenderPhase + i * renderSliceSize; + + OwnPtr<ReverbConvolverStage> stage(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer)); + + bool isBackgroundStage = false; + + if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) { + m_backgroundStages.append(stage.release()); + isBackgroundStage = true; + } else + m_stages.append(stage.release()); + + stageOffset += stageSize; + ++i; + + // Figure out next FFT size + fftSize *= 2; + if (hasRealtimeConstraint && !isBackgroundStage && fftSize > m_maxRealtimeFFTSize) + fftSize = m_maxRealtimeFFTSize; + if (fftSize > m_maxFFTSize) + fftSize = m_maxFFTSize; + } + + // 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.size() > 0) + m_backgroundThread = createThread(WebCore::backgroundThreadEntry, this, "convolution background thread"); +} + +ReverbConvolver::~ReverbConvolver() +{ + // Wait for background thread to stop + if (useBackgroundThreads() && m_backgroundThread) { + m_wantsToExit = true; + + // Wake up thread so it can return + { + MutexLocker locker(m_backgroundThreadLock); + m_moreInputBuffered = true; + m_backgroundThreadCondition.signal(); + } + + waitForThreadCompletion(m_backgroundThread, 0); + } +} + +void ReverbConvolver::backgroundThreadEntry() +{ + while (!m_wantsToExit) { + // Wait for realtime thread to give us more input + m_moreInputBuffered = false; + { + MutexLocker locker(m_backgroundThreadLock); + while (!m_moreInputBuffered && !m_wantsToExit) + m_backgroundThreadCondition.wait(m_backgroundThreadLock); + } + + // 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... + // The ReverbConvolverStages need to process in amounts which evenly divide half the FFT size + const int SliceSize = MinFFTSize / 2; + + // Accumulate contributions from each stage + for (size_t i = 0; i < m_backgroundStages.size(); ++i) + m_backgroundStages[i]->processInBackground(this, SliceSize); + } + } +} + +void ReverbConvolver::process(AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess) +{ + bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess; + ASSERT(isSafe); + if (!isSafe) + return; + + float* source = sourceChannel->data(); + float* destination = destinationChannel->data(); + bool isDataSafe = source && destination; + ASSERT(isDataSafe); + if (!isDataSafe) + return; + + // Feed input buffer (read by all threads) + m_inputBuffer.write(source, framesToProcess); + + // Accumulate contributions from each stage + for (size_t i = 0; i < m_stages.size(); ++i) + m_stages[i]->process(source, framesToProcess); + + // Finally read from accumulation buffer + m_accumulationBuffer.readAndClear(destination, framesToProcess); + + // 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.tryLock()) { + m_moreInputBuffered = true; + m_backgroundThreadCondition.signal(); + m_backgroundThreadLock.unlock(); + } +} + +void ReverbConvolver::reset() +{ + for (size_t i = 0; i < m_stages.size(); ++i) + m_stages[i]->reset(); + + for (size_t i = 0; i < m_backgroundStages.size(); ++i) + m_backgroundStages[i]->reset(); + + m_accumulationBuffer.reset(); + m_inputBuffer.reset(); +} + +} // namespace WebCore + +#endif // ENABLE(WEB_AUDIO) |