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Diffstat (limited to 'WebCore/platform/audio/ReverbConvolver.cpp')
| -rw-r--r-- | WebCore/platform/audio/ReverbConvolver.cpp | 230 |
1 files changed, 0 insertions, 230 deletions
diff --git a/WebCore/platform/audio/ReverbConvolver.cpp b/WebCore/platform/audio/ReverbConvolver.cpp deleted file mode 100644 index 120c9f2..0000000 --- a/WebCore/platform/audio/ReverbConvolver.cpp +++ /dev/null @@ -1,230 +0,0 @@ -/* - * 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) |