third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp - chromium/src - 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.
  *
  * THIS SOFTWARE IS PROVIDED BY APPLE INC. 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 INC. 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 "bindings/core/v8/ExceptionState.h"
 #include "core/dom/ExceptionCode.h"
 #include "modules/webaudio/AudioBuffer.h"
 #include "modules/webaudio/AudioNodeInput.h"
 #include "modules/webaudio/AudioNodeOutput.h"
 #include "modules/webaudio/ConvolverNode.h"
 #include "modules/webaudio/ConvolverOptions.h"
 #include "platform/audio/Reverb.h"
 #include "wtf/PtrUtil.h"
 #include <memory>

 // Note about empirical tuning:
 // The maximum FFT size affects reverb performance and accuracy.
 // If the reverb is single-threaded and processes entirely in the real-time
 // audio thread, it's important not to make this too high.  In this case 8192 is
 // a good value.  But, the Reverb object is multi-threaded, so we want this as
 // high as possible without losing too much accuracy.  Very large FFTs will have
 // worse phase errors. Given these constraints 32768 is a good compromise.
 const size_t MaxFFTSize = 32768;

 namespace blink {

 ConvolverHandler::ConvolverHandler(AudioNode& node, float sampleRate)
     : AudioHandler(NodeTypeConvolver, node, sampleRate), m_normalize(true) {
   addInput();
   addOutput(2);

   // Node-specific default mixing rules.
   m_channelCount = 2;
   setInternalChannelCountMode(ClampedMax);
   setInternalChannelInterpretation(AudioBus::Speakers);

   initialize();
 }

 PassRefPtr<ConvolverHandler> ConvolverHandler::create(AudioNode& node,
                                                       float sampleRate) {
   return adoptRef(new ConvolverHandler(node, sampleRate));
 }

 ConvolverHandler::~ConvolverHandler() {
   uninitialize();
 }

 void ConvolverHandler::process(size_t framesToProcess) {
   AudioBus* outputBus = output(0).bus();
   DCHECK(outputBus);

   // Synchronize with possible dynamic changes to the impulse response.
   MutexTryLocker tryLocker(m_processLock);
   if (tryLocker.locked()) {
     if (!isInitialized() || !m_reverb) {
       outputBus->zero();
     } else {
       // Process using the convolution engine.
       // Note that we can handle the case where nothing is connected to the
       // input, in which case we'll just feed silence into the convolver.
       // FIXME:  If we wanted to get fancy we could try to factor in the 'tail
       // time' and stop processing once the tail dies down if
       // we keep getting fed silence.
       m_reverb->process(input(0).bus(), outputBus, framesToProcess);
     }
   } else {
     // Too bad - the tryLock() failed.  We must be in the middle of setting a
     // new impulse response.
     outputBus->zero();
   }
 }

 void ConvolverHandler::setBuffer(AudioBuffer* buffer,
                                  ExceptionState& exceptionState) {
   DCHECK(isMainThread());

   if (!buffer)
     return;

   if (buffer->sampleRate() != context()->sampleRate()) {
     exceptionState.throwDOMException(
         NotSupportedError,
         "The buffer sample rate of " + String::number(buffer->sampleRate()) +
             " does not match the context rate of " +
             String::number(context()->sampleRate()) + " Hz.");
     return;
   }

   unsigned numberOfChannels = buffer->numberOfChannels();
   size_t bufferLength = buffer->length();

   // The current implementation supports only 1-, 2-, or 4-channel impulse
   // responses, with the 4-channel response being interpreted as true-stereo
   // (see Reverb class).
   bool isChannelCountGood =
       numberOfChannels == 1 || numberOfChannels == 2 || numberOfChannels == 4;

   if (!isChannelCountGood) {
     exceptionState.throwDOMException(
         NotSupportedError, "The buffer must have 1, 2, or 4 channels, not " +
                                String::number(numberOfChannels));
     return;
   }

   // Wrap the AudioBuffer by an AudioBus. It's an efficient pointer set and not
   // a memcpy().  This memory is simply used in the Reverb constructor and no
   // reference to it is kept for later use in that class.
   RefPtr<AudioBus> bufferBus =
       AudioBus::create(numberOfChannels, bufferLength, false);
   for (unsigned i = 0; i < numberOfChannels; ++i)
     bufferBus->setChannelMemory(i, buffer->getChannelData(i)->data(),
                                 bufferLength);

   bufferBus->setSampleRate(buffer->sampleRate());

   // Create the reverb with the given impulse response.
   std::unique_ptr<Reverb> reverb = wrapUnique(
       new Reverb(bufferBus.get(), ProcessingSizeInFrames, MaxFFTSize, 2,
                  context() && context()->hasRealtimeConstraint(), m_normalize));

   {
     // Synchronize with process().
     MutexLocker locker(m_processLock);
     m_reverb = std::move(reverb);
     m_buffer = buffer;
   }
 }

 AudioBuffer* ConvolverHandler::buffer() {
   DCHECK(isMainThread());
   return m_buffer.get();
 }

 double ConvolverHandler::tailTime() const {
   MutexTryLocker tryLocker(m_processLock);
   if (tryLocker.locked())
     return m_reverb
                ? m_reverb->impulseResponseLength() /
                      static_cast<double>(sampleRate())
                : 0;
   // Since we don't want to block the Audio Device thread, we return a large
   // value instead of trying to acquire the lock.
   return std::numeric_limits<double>::infinity();
 }

 double ConvolverHandler::latencyTime() const {
   MutexTryLocker tryLocker(m_processLock);
   if (tryLocker.locked())
     return m_reverb
                ? m_reverb->latencyFrames() / static_cast<double>(sampleRate())
                : 0;
   // Since we don't want to block the Audio Device thread, we return a large
   // value instead of trying to acquire the lock.
   return std::numeric_limits<double>::infinity();
 }

 // ----------------------------------------------------------------

 ConvolverNode::ConvolverNode(BaseAudioContext& context) : AudioNode(context) {
   setHandler(ConvolverHandler::create(*this, context.sampleRate()));
 }

 ConvolverNode* ConvolverNode::create(BaseAudioContext& context,
                                      ExceptionState& exceptionState) {
   DCHECK(isMainThread());

   if (context.isContextClosed()) {
     context.throwExceptionForClosedState(exceptionState);
     return nullptr;
   }

   return new ConvolverNode(context);
 }

 ConvolverNode* ConvolverNode::create(BaseAudioContext* context,
                                      const ConvolverOptions& options,
                                      ExceptionState& exceptionState) {
   ConvolverNode* node = create(*context, exceptionState);

   if (!node)
     return nullptr;

   node->handleChannelOptions(options, exceptionState);

   // It is important to set normalize first because setting the buffer will
   // examing the normalize attribute to see if normalization needs to be done.
   node->setNormalize(!options.disableNormalization());
   if (options.hasBuffer())
     node->setBuffer(options.buffer(), exceptionState);
   return node;
 }

 ConvolverHandler& ConvolverNode::convolverHandler() const {
   return static_cast<ConvolverHandler&>(handler());
 }

 AudioBuffer* ConvolverNode::buffer() const {
   return convolverHandler().buffer();
 }

 void ConvolverNode::setBuffer(AudioBuffer* newBuffer,
                               ExceptionState& exceptionState) {
   convolverHandler().setBuffer(newBuffer, exceptionState);
 }

 bool ConvolverNode::normalize() const {
   return convolverHandler().normalize();
 }

 void ConvolverNode::setNormalize(bool normalize) {
   convolverHandler().setNormalize(normalize);
 }

 }  // namespace blink
	/*
	* 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.
	*
	* THIS SOFTWARE IS PROVIDED BY APPLE INC. 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 INC. 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 "bindings/core/v8/ExceptionState.h"
	#include "core/dom/ExceptionCode.h"
	#include "modules/webaudio/AudioBuffer.h"
	#include "modules/webaudio/AudioNodeInput.h"
	#include "modules/webaudio/AudioNodeOutput.h"
	#include "modules/webaudio/ConvolverNode.h"
	#include "modules/webaudio/ConvolverOptions.h"
	#include "platform/audio/Reverb.h"
	#include "wtf/PtrUtil.h"
	#include <memory>

	// Note about empirical tuning:
	// The maximum FFT size affects reverb performance and accuracy.
	// If the reverb is single-threaded and processes entirely in the real-time
	// audio thread, it's important not to make this too high. In this case 8192 is
	// a good value. But, the Reverb object is multi-threaded, so we want this as
	// high as possible without losing too much accuracy. Very large FFTs will have
	// worse phase errors. Given these constraints 32768 is a good compromise.
	const size_t MaxFFTSize = 32768;

	namespace blink {

	ConvolverHandler::ConvolverHandler(AudioNode& node, float sampleRate)
	: AudioHandler(NodeTypeConvolver, node, sampleRate), m_normalize(true) {
	addInput();
	addOutput(2);

	// Node-specific default mixing rules.
	m_channelCount = 2;
	setInternalChannelCountMode(ClampedMax);
	setInternalChannelInterpretation(AudioBus::Speakers);

	initialize();
	}

	PassRefPtr<ConvolverHandler> ConvolverHandler::create(AudioNode& node,
	float sampleRate) {
	return adoptRef(new ConvolverHandler(node, sampleRate));
	}

	ConvolverHandler::~ConvolverHandler() {
	uninitialize();
	}

	void ConvolverHandler::process(size_t framesToProcess) {
	AudioBus* outputBus = output(0).bus();
	DCHECK(outputBus);

	// Synchronize with possible dynamic changes to the impulse response.
	MutexTryLocker tryLocker(m_processLock);
	if (tryLocker.locked()) {
	if (!isInitialized() \|\| !m_reverb) {
	outputBus->zero();
	} else {
	// Process using the convolution engine.
	// Note that we can handle the case where nothing is connected to the
	// input, in which case we'll just feed silence into the convolver.
	// FIXME: If we wanted to get fancy we could try to factor in the 'tail
	// time' and stop processing once the tail dies down if
	// we keep getting fed silence.
	m_reverb->process(input(0).bus(), outputBus, framesToProcess);
	}
	} else {
	// Too bad - the tryLock() failed. We must be in the middle of setting a
	// new impulse response.
	outputBus->zero();
	}
	}

	void ConvolverHandler::setBuffer(AudioBuffer* buffer,
	ExceptionState& exceptionState) {
	DCHECK(isMainThread());

	if (!buffer)
	return;

	if (buffer->sampleRate() != context()->sampleRate()) {
	exceptionState.throwDOMException(
	NotSupportedError,
	"The buffer sample rate of " + String::number(buffer->sampleRate()) +
	" does not match the context rate of " +
	String::number(context()->sampleRate()) + " Hz.");
	return;
	}

	unsigned numberOfChannels = buffer->numberOfChannels();
	size_t bufferLength = buffer->length();

	// The current implementation supports only 1-, 2-, or 4-channel impulse
	// responses, with the 4-channel response being interpreted as true-stereo
	// (see Reverb class).
	bool isChannelCountGood =
	numberOfChannels == 1 \|\| numberOfChannels == 2 \|\| numberOfChannels == 4;

	if (!isChannelCountGood) {
	exceptionState.throwDOMException(
	NotSupportedError, "The buffer must have 1, 2, or 4 channels, not " +
	String::number(numberOfChannels));
	return;
	}

	// Wrap the AudioBuffer by an AudioBus. It's an efficient pointer set and not
	// a memcpy(). This memory is simply used in the Reverb constructor and no
	// reference to it is kept for later use in that class.
	RefPtr<AudioBus> bufferBus =
	AudioBus::create(numberOfChannels, bufferLength, false);
	for (unsigned i = 0; i < numberOfChannels; ++i)
	bufferBus->setChannelMemory(i, buffer->getChannelData(i)->data(),
	bufferLength);

	bufferBus->setSampleRate(buffer->sampleRate());

	// Create the reverb with the given impulse response.
	std::unique_ptr<Reverb> reverb = wrapUnique(
	new Reverb(bufferBus.get(), ProcessingSizeInFrames, MaxFFTSize, 2,
	context() && context()->hasRealtimeConstraint(), m_normalize));

	{
	// Synchronize with process().
	MutexLocker locker(m_processLock);
	m_reverb = std::move(reverb);
	m_buffer = buffer;
	}
	}

	AudioBuffer* ConvolverHandler::buffer() {
	DCHECK(isMainThread());
	return m_buffer.get();
	}

	double ConvolverHandler::tailTime() const {
	MutexTryLocker tryLocker(m_processLock);
	if (tryLocker.locked())
	return m_reverb
	? m_reverb->impulseResponseLength() /
	static_cast<double>(sampleRate())
	: 0;
	// Since we don't want to block the Audio Device thread, we return a large
	// value instead of trying to acquire the lock.
	return std::numeric_limits<double>::infinity();
	}

	double ConvolverHandler::latencyTime() const {
	MutexTryLocker tryLocker(m_processLock);
	if (tryLocker.locked())
	return m_reverb
	? m_reverb->latencyFrames() / static_cast<double>(sampleRate())
	: 0;
	// Since we don't want to block the Audio Device thread, we return a large
	// value instead of trying to acquire the lock.
	return std::numeric_limits<double>::infinity();
	}

	// ----------------------------------------------------------------

	ConvolverNode::ConvolverNode(BaseAudioContext& context) : AudioNode(context) {
	setHandler(ConvolverHandler::create(*this, context.sampleRate()));
	}

	ConvolverNode* ConvolverNode::create(BaseAudioContext& context,
	ExceptionState& exceptionState) {
	DCHECK(isMainThread());

	if (context.isContextClosed()) {
	context.throwExceptionForClosedState(exceptionState);
	return nullptr;
	}

	return new ConvolverNode(context);
	}

	ConvolverNode* ConvolverNode::create(BaseAudioContext* context,
	const ConvolverOptions& options,
	ExceptionState& exceptionState) {
	ConvolverNode* node = create(*context, exceptionState);

	if (!node)
	return nullptr;

	node->handleChannelOptions(options, exceptionState);

	// It is important to set normalize first because setting the buffer will
	// examing the normalize attribute to see if normalization needs to be done.
	node->setNormalize(!options.disableNormalization());
	if (options.hasBuffer())
	node->setBuffer(options.buffer(), exceptionState);
	return node;
	}

	ConvolverHandler& ConvolverNode::convolverHandler() const {
	return static_cast<ConvolverHandler&>(handler());
	}

	AudioBuffer* ConvolverNode::buffer() const {
	return convolverHandler().buffer();
	}

	void ConvolverNode::setBuffer(AudioBuffer* newBuffer,
	ExceptionState& exceptionState) {
	convolverHandler().setBuffer(newBuffer, exceptionState);
	}

	bool ConvolverNode::normalize() const {
	return convolverHandler().normalize();
	}

	void ConvolverNode::setNormalize(bool normalize) {
	convolverHandler().setNormalize(normalize);
	}

	} // namespace blink