| /* |
| * 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 |
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| * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #include "config.h" |
| |
| #if ENABLE(WEB_AUDIO) |
| |
| #include "platform/audio/ReverbConvolver.h" |
| |
| #include "platform/audio/AudioBus.h" |
| #include "platform/audio/VectorMath.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_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) |