Source/WebCore/platform/audio/ReverbConvolver.cpp - WebKit - Git at Google

 /*
  * 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 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"
 #include <mutex>

 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;

 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)
 {
     // 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;

         auto stage = makeUnique<ReverbConvolverStage>(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver);

         bool isBackgroundStage = false;

         if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
             m_backgroundStages.append(WTFMove(stage));
             isBackgroundStage = true;
         } else
             m_stages.append(WTFMove(stage));

         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 = Thread::create("convolution background thread", [this] {
             backgroundThreadEntry();
         }, ThreadType::Audio);
     }
 }

 ReverbConvolver::~ReverbConvolver()
 {
     // Wait for background thread to stop
     if (useBackgroundThreads() && m_backgroundThread) {
         m_wantsToExit = true;

         // Wake up thread so it can return
         {
             auto locker = holdLock(m_backgroundThreadMutex);
             m_moreInputBuffered = true;
             m_backgroundThreadConditionVariable.notifyOne();
         }

         m_backgroundThread->waitForCompletion();
     }
 }

 void ReverbConvolver::backgroundThreadEntry()
 {
     while (!m_wantsToExit) {
         // Wait for realtime thread to give us more input
         m_moreInputBuffered = false;
         {
             std::unique_lock<Lock> lock(m_backgroundThreadMutex);

             m_backgroundThreadConditionVariable.wait(lock, [this] { return m_moreInputBuffered || m_wantsToExit; });
         }

         // 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.

     // We use use std::unique_lock with std::try_lock here 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...
     std::unique_lock<Lock> lock(m_backgroundThreadMutex, std::try_to_lock);
     if (!lock.owns_lock())
         return;

     m_moreInputBuffered = true;
     m_backgroundThreadConditionVariable.notifyOne();
 }

 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)
	/*
	* 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 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"
	#include <mutex>

	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;

	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)
	{
	// 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;

	auto stage = makeUnique<ReverbConvolverStage>(response, totalResponseLength, reverbTotalLatency, stageOffset, stageSize, fftSize, renderPhase, renderSliceSize, &m_accumulationBuffer, useDirectConvolver);

	bool isBackgroundStage = false;

	if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
	m_backgroundStages.append(WTFMove(stage));
	isBackgroundStage = true;
	} else
	m_stages.append(WTFMove(stage));

	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 = Thread::create("convolution background thread", [this] {
	backgroundThreadEntry();
	}, ThreadType::Audio);
	}
	}

	ReverbConvolver::~ReverbConvolver()
	{
	// Wait for background thread to stop
	if (useBackgroundThreads() && m_backgroundThread) {
	m_wantsToExit = true;

	// Wake up thread so it can return
	{
	auto locker = holdLock(m_backgroundThreadMutex);
	m_moreInputBuffered = true;
	m_backgroundThreadConditionVariable.notifyOne();
	}

	m_backgroundThread->waitForCompletion();
	}
	}

	void ReverbConvolver::backgroundThreadEntry()
	{
	while (!m_wantsToExit) {
	// Wait for realtime thread to give us more input
	m_moreInputBuffered = false;
	{
	std::unique_lock<Lock> lock(m_backgroundThreadMutex);

	m_backgroundThreadConditionVariable.wait(lock, [this] { return m_moreInputBuffered \|\| m_wantsToExit; });
	}

	// 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.

	// We use use std::unique_lock with std::try_lock here 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...
	std::unique_lock<Lock> lock(m_backgroundThreadMutex, std::try_to_lock);
	if (!lock.owns_lock())
	return;

	m_moreInputBuffered = true;
	m_backgroundThreadConditionVariable.notifyOne();
	}

	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)