/* * 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 = 128; const size_t MaxRealtimeFFTSize = 2048; static void backgroundThreadEntry(void* threadData) { ReverbConvolver* reverbConvolver = static_cast<ReverbConvolver*>(threadData); reverbConvolver->backgroundThreadEntry(); } 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; const float* response = impulseResponse->data(); size_t totalResponseLength = impulseResponse->length(); // The total latency is zero because the direct-convolution is used in the leading portion. size_t reverbTotalLatency = 0; 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; bool useDirectConvolver = !stageOffset; OwnPtr<ReverbConvolverStage> stage = adoptPtr(new ReverbConvolverStage(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver)); bool isBackgroundStage = false; if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) { m_backgroundStages.append(stage.release()); isBackgroundStage = true; } else m_stages.append(stage.release()); stageOffset += stageSize; ++i; if (!useDirectConvolver) { // 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); } } 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(const AudioChannel* sourceChannel, AudioChannel* destinationChannel, size_t framesToProcess) { bool isSafe = sourceChannel && destinationChannel && sourceChannel->length() >= framesToProcess && destinationChannel->length() >= framesToProcess; ASSERT(isSafe); if (!isSafe) return; const float* source = sourceChannel->data(); float* destination = destinationChannel->mutableData(); 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(); } size_t ReverbConvolver::latencyFrames() const { return 0; } } // namespace WebCore #endif // ENABLE(WEB_AUDIO)