Source/WebCore/Modules/webaudio/ScriptProcessorNode.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.
  *
  * 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 "config.h"

 #if ENABLE(WEB_AUDIO)

 #include "ScriptProcessorNode.h"

 #include "AudioBuffer.h"
 #include "AudioBus.h"
 #include "AudioContext.h"
 #include "AudioNodeInput.h"
 #include "AudioNodeOutput.h"
 #include "AudioProcessingEvent.h"
 #include "Document.h"
 #include "EventNames.h"
 #include <JavaScriptCore/Float32Array.h>
 #include <wtf/IsoMallocInlines.h>
 #include <wtf/MainThread.h>

 namespace WebCore {

 WTF_MAKE_ISO_ALLOCATED_IMPL(ScriptProcessorNode);

 Ref<ScriptProcessorNode> ScriptProcessorNode::create(BaseAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
 {
     return adoptRef(*new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels));
 }

 ScriptProcessorNode::ScriptProcessorNode(BaseAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
     : AudioNode(context, sampleRate)
     , ActiveDOMObject(context.scriptExecutionContext())
     , m_doubleBufferIndex(0)
     , m_doubleBufferIndexForEvent(0)
     , m_bufferSize(bufferSize)
     , m_bufferReadWriteIndex(0)
     , m_isRequestOutstanding(false)
     , m_numberOfInputChannels(numberOfInputChannels)
     , m_numberOfOutputChannels(numberOfOutputChannels)
     , m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false))
 {
     // Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode.
     if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
         m_bufferSize = AudioNode::ProcessingSizeInFrames;

     ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels());

     setNodeType(NodeTypeJavaScript);
     addInput(makeUnique<AudioNodeInput>(this));
     addOutput(makeUnique<AudioNodeOutput>(this, numberOfOutputChannels));

     initialize();
     suspendIfNeeded();
     m_pendingActivity = makePendingActivity(*this);
 }

 ScriptProcessorNode::~ScriptProcessorNode()
 {
     ASSERT(!hasPendingActivity());
     uninitialize();
 }

 void ScriptProcessorNode::initialize()
 {
     if (isInitialized())
         return;

     float sampleRate = context().sampleRate();

     // Create double buffers on both the input and output sides.
     // These AudioBuffers will be directly accessed in the main thread by JavaScript.
     for (unsigned i = 0; i < 2; ++i) {
         auto inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0;
         auto outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;

         m_inputBuffers.append(inputBuffer);
         m_outputBuffers.append(outputBuffer);
     }

     AudioNode::initialize();
 }

 void ScriptProcessorNode::uninitialize()
 {
     if (!isInitialized())
         return;

     m_inputBuffers.clear();
     m_outputBuffers.clear();

     AudioNode::uninitialize();
 }

 void ScriptProcessorNode::didBecomeMarkedForDeletion()
 {
     ASSERT(context().isGraphOwner());
     m_pendingActivity = nullptr;
     ASSERT(!hasPendingActivity());
 }

 void ScriptProcessorNode::process(size_t framesToProcess)
 {
     // Discussion about inputs and outputs:
     // As in other AudioNodes, ScriptProcessorNode uses an AudioBus for its input and output (see inputBus and outputBus below).
     // Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
     // This node is the producer for inputBuffer and the consumer for outputBuffer.
     // The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.

     // Get input and output busses.
     AudioBus* inputBus = this->input(0)->bus();
     AudioBus* outputBus = this->output(0)->bus();

     // Get input and output buffers. We double-buffer both the input and output sides.
     unsigned doubleBufferIndex = this->doubleBufferIndex();
     bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
     ASSERT(isDoubleBufferIndexGood);
     if (!isDoubleBufferIndexGood)
         return;

     AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
     AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();

     // Check the consistency of input and output buffers.
     unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
     bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();

     // If the number of input channels is zero, it's ok to have inputBuffer = 0.
     if (m_internalInputBus->numberOfChannels())
         buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();

     ASSERT(buffersAreGood);
     if (!buffersAreGood)
         return;

     // We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
     bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
     ASSERT(isFramesToProcessGood);
     if (!isFramesToProcessGood)
         return;

     unsigned numberOfOutputChannels = outputBus->numberOfChannels();

     bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
     ASSERT(channelsAreGood);
     if (!channelsAreGood)
         return;

     for (unsigned i = 0; i < numberOfInputChannels; i++)
         m_internalInputBus->setChannelMemory(i, inputBuffer->channelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);

     if (numberOfInputChannels)
         m_internalInputBus->copyFrom(*inputBus);

     // Copy from the output buffer to the output.
     for (unsigned i = 0; i < numberOfOutputChannels; ++i)
         memcpy(outputBus->channel(i)->mutableData(), outputBuffer->channelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);

     // Update the buffering index.
     m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();

     // m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
     // When this happens, fire an event and swap buffers.
     if (!m_bufferReadWriteIndex) {
         // Avoid building up requests on the main thread to fire process events when they're not being handled.
         // This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
         if (m_isRequestOutstanding) {
             // We're late in handling the previous request. The main thread must be very busy.
             // The best we can do is clear out the buffer ourself here.
             outputBuffer->zero();
         } else {
             // Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
             ref();

             // Fire the event on the main thread, not this one (which is the realtime audio thread).
             m_doubleBufferIndexForEvent = m_doubleBufferIndex;
             m_isRequestOutstanding = true;

             callOnMainThread([this] {
                 fireProcessEvent();

                 // De-reference to match the ref() call in process().
                 deref();
             });
         }

         swapBuffers();
     }
 }

 void ScriptProcessorNode::fireProcessEvent()
 {
     ASSERT(isMainThread() && m_isRequestOutstanding);

     bool isIndexGood = m_doubleBufferIndexForEvent < 2;
     ASSERT(isIndexGood);
     if (!isIndexGood)
         return;

     AudioBuffer* inputBuffer = m_inputBuffers[m_doubleBufferIndexForEvent].get();
     AudioBuffer* outputBuffer = m_outputBuffers[m_doubleBufferIndexForEvent].get();
     ASSERT(outputBuffer);
     if (!outputBuffer)
         return;

     // Avoid firing the event if the document has already gone away.
     if (!context().isStopped()) {
         // Let the audio thread know we've gotten to the point where it's OK for it to make another request.
         m_isRequestOutstanding = false;

         // Calculate playbackTime with the buffersize which needs to be processed each time when onaudioprocess is called.
         // The outputBuffer being passed to JS will be played after exhausting previous outputBuffer by double-buffering.
         double playbackTime = (context().currentSampleFrame() + m_bufferSize) / static_cast<double>(context().sampleRate());

         // Call the JavaScript event handler which will do the audio processing.
         dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime));
     }
 }

 void ScriptProcessorNode::reset()
 {
     m_bufferReadWriteIndex = 0;
     m_doubleBufferIndex = 0;

     for (unsigned i = 0; i < 2; ++i) {
         m_inputBuffers[i]->zero();
         m_outputBuffers[i]->zero();
     }
 }

 double ScriptProcessorNode::tailTime() const
 {
     return std::numeric_limits<double>::infinity();
 }

 double ScriptProcessorNode::latencyTime() const
 {
     return std::numeric_limits<double>::infinity();
 }

 } // 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.
	*
	* 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 "config.h"

	#if ENABLE(WEB_AUDIO)

	#include "ScriptProcessorNode.h"

	#include "AudioBuffer.h"
	#include "AudioBus.h"
	#include "AudioContext.h"
	#include "AudioNodeInput.h"
	#include "AudioNodeOutput.h"
	#include "AudioProcessingEvent.h"
	#include "Document.h"
	#include "EventNames.h"
	#include <JavaScriptCore/Float32Array.h>
	#include <wtf/IsoMallocInlines.h>
	#include <wtf/MainThread.h>

	namespace WebCore {

	WTF_MAKE_ISO_ALLOCATED_IMPL(ScriptProcessorNode);

	Ref<ScriptProcessorNode> ScriptProcessorNode::create(BaseAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
	{
	return adoptRef(*new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels));
	}

	ScriptProcessorNode::ScriptProcessorNode(BaseAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
	: AudioNode(context, sampleRate)
	, ActiveDOMObject(context.scriptExecutionContext())
	, m_doubleBufferIndex(0)
	, m_doubleBufferIndexForEvent(0)
	, m_bufferSize(bufferSize)
	, m_bufferReadWriteIndex(0)
	, m_isRequestOutstanding(false)
	, m_numberOfInputChannels(numberOfInputChannels)
	, m_numberOfOutputChannels(numberOfOutputChannels)
	, m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false))
	{
	// Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode.
	if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
	m_bufferSize = AudioNode::ProcessingSizeInFrames;

	ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels());

	setNodeType(NodeTypeJavaScript);
	addInput(makeUnique<AudioNodeInput>(this));
	addOutput(makeUnique<AudioNodeOutput>(this, numberOfOutputChannels));

	initialize();
	suspendIfNeeded();
	m_pendingActivity = makePendingActivity(*this);
	}

	ScriptProcessorNode::~ScriptProcessorNode()
	{
	ASSERT(!hasPendingActivity());
	uninitialize();
	}

	void ScriptProcessorNode::initialize()
	{
	if (isInitialized())
	return;

	float sampleRate = context().sampleRate();

	// Create double buffers on both the input and output sides.
	// These AudioBuffers will be directly accessed in the main thread by JavaScript.
	for (unsigned i = 0; i < 2; ++i) {
	auto inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0;
	auto outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;

	m_inputBuffers.append(inputBuffer);
	m_outputBuffers.append(outputBuffer);
	}

	AudioNode::initialize();
	}

	void ScriptProcessorNode::uninitialize()
	{
	if (!isInitialized())
	return;

	m_inputBuffers.clear();
	m_outputBuffers.clear();

	AudioNode::uninitialize();
	}

	void ScriptProcessorNode::didBecomeMarkedForDeletion()
	{
	ASSERT(context().isGraphOwner());
	m_pendingActivity = nullptr;
	ASSERT(!hasPendingActivity());
	}

	void ScriptProcessorNode::process(size_t framesToProcess)
	{
	// Discussion about inputs and outputs:
	// As in other AudioNodes, ScriptProcessorNode uses an AudioBus for its input and output (see inputBus and outputBus below).
	// Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
	// This node is the producer for inputBuffer and the consumer for outputBuffer.
	// The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.

	// Get input and output busses.
	AudioBus* inputBus = this->input(0)->bus();
	AudioBus* outputBus = this->output(0)->bus();

	// Get input and output buffers. We double-buffer both the input and output sides.
	unsigned doubleBufferIndex = this->doubleBufferIndex();
	bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
	ASSERT(isDoubleBufferIndexGood);
	if (!isDoubleBufferIndexGood)
	return;

	AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
	AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();

	// Check the consistency of input and output buffers.
	unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
	bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();

	// If the number of input channels is zero, it's ok to have inputBuffer = 0.
	if (m_internalInputBus->numberOfChannels())
	buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();

	ASSERT(buffersAreGood);
	if (!buffersAreGood)
	return;

	// We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
	bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
	ASSERT(isFramesToProcessGood);
	if (!isFramesToProcessGood)
	return;

	unsigned numberOfOutputChannels = outputBus->numberOfChannels();

	bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
	ASSERT(channelsAreGood);
	if (!channelsAreGood)
	return;

	for (unsigned i = 0; i < numberOfInputChannels; i++)
	m_internalInputBus->setChannelMemory(i, inputBuffer->channelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);

	if (numberOfInputChannels)
	m_internalInputBus->copyFrom(*inputBus);

	// Copy from the output buffer to the output.
	for (unsigned i = 0; i < numberOfOutputChannels; ++i)
	memcpy(outputBus->channel(i)->mutableData(), outputBuffer->channelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);

	// Update the buffering index.
	m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();

	// m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
	// When this happens, fire an event and swap buffers.
	if (!m_bufferReadWriteIndex) {
	// Avoid building up requests on the main thread to fire process events when they're not being handled.
	// This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
	if (m_isRequestOutstanding) {
	// We're late in handling the previous request. The main thread must be very busy.
	// The best we can do is clear out the buffer ourself here.
	outputBuffer->zero();
	} else {
	// Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
	ref();

	// Fire the event on the main thread, not this one (which is the realtime audio thread).
	m_doubleBufferIndexForEvent = m_doubleBufferIndex;
	m_isRequestOutstanding = true;

	callOnMainThread([this] {
	fireProcessEvent();

	// De-reference to match the ref() call in process().
	deref();
	});
	}

	swapBuffers();
	}
	}

	void ScriptProcessorNode::fireProcessEvent()
	{
	ASSERT(isMainThread() && m_isRequestOutstanding);

	bool isIndexGood = m_doubleBufferIndexForEvent < 2;
	ASSERT(isIndexGood);
	if (!isIndexGood)
	return;

	AudioBuffer* inputBuffer = m_inputBuffers[m_doubleBufferIndexForEvent].get();
	AudioBuffer* outputBuffer = m_outputBuffers[m_doubleBufferIndexForEvent].get();
	ASSERT(outputBuffer);
	if (!outputBuffer)
	return;

	// Avoid firing the event if the document has already gone away.
	if (!context().isStopped()) {
	// Let the audio thread know we've gotten to the point where it's OK for it to make another request.
	m_isRequestOutstanding = false;

	// Calculate playbackTime with the buffersize which needs to be processed each time when onaudioprocess is called.
	// The outputBuffer being passed to JS will be played after exhausting previous outputBuffer by double-buffering.
	double playbackTime = (context().currentSampleFrame() + m_bufferSize) / static_cast<double>(context().sampleRate());

	// Call the JavaScript event handler which will do the audio processing.
	dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime));
	}
	}

	void ScriptProcessorNode::reset()
	{
	m_bufferReadWriteIndex = 0;
	m_doubleBufferIndex = 0;

	for (unsigned i = 0; i < 2; ++i) {
	m_inputBuffers[i]->zero();
	m_outputBuffers[i]->zero();
	}
	}

	double ScriptProcessorNode::tailTime() const
	{
	return std::numeric_limits<double>::infinity();
	}

	double ScriptProcessorNode::latencyTime() const
	{
	return std::numeric_limits<double>::infinity();
	}

	} // namespace WebCore

	#endif // ENABLE(WEB_AUDIO)