Source/WebCore/webaudio/JavaScriptAudioNode.cpp - platform/external/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 "JavaScriptAudioNode.h"

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

 namespace WebCore {

 const size_t DefaultBufferSize = 4096;

 PassRefPtr<JavaScriptAudioNode> JavaScriptAudioNode::create(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
 {
     return adoptRef(new JavaScriptAudioNode(context, sampleRate, bufferSize, numberOfInputs, numberOfOutputs));
 }

 JavaScriptAudioNode::JavaScriptAudioNode(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
     : AudioNode(context, sampleRate)
     , m_doubleBufferIndex(0)
     , m_doubleBufferIndexForEvent(0)
     , m_bufferSize(bufferSize)
     , m_bufferReadWriteIndex(0)
     , m_isRequestOutstanding(false)
 {
     // Check for valid buffer size.
     switch (bufferSize) {
     case 256:
     case 512:
     case 1024:
     case 2048:
     case 4096:
     case 8192:
     case 16384:
         m_bufferSize = bufferSize;
         break;
     default:
         m_bufferSize = DefaultBufferSize;
     }

     // Regardless of the allowed buffer sizes above, we still need to process at the granularity of the AudioNode.
     if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
         m_bufferSize = AudioNode::ProcessingSizeInFrames;

     // FIXME: Right now we're hardcoded to single input and single output.
     // Although the specification says this is OK for a simple implementation, multiple inputs and outputs would be good.
     ASSERT_UNUSED(numberOfInputs, numberOfInputs == 1);
     ASSERT_UNUSED(numberOfOutputs, numberOfOutputs == 1);
     addInput(adoptPtr(new AudioNodeInput(this)));
     addOutput(adoptPtr(new AudioNodeOutput(this, 2)));

     setType(NodeTypeJavaScript);

     initialize();
 }

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

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

     double 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) {
         m_inputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
         m_outputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
     }

     AudioNode::initialize();
 }

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

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

     AudioNode::uninitialize();
 }

 JavaScriptAudioNode* JavaScriptAudioNode::toJavaScriptAudioNode()
 {
     return this;
 }

 void JavaScriptAudioNode::process(size_t framesToProcess)
 {
     // Discussion about inputs and outputs:
     // As in other AudioNodes, JavaScriptAudioNode 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.
     bool buffersAreGood = inputBuffer && outputBuffer && bufferSize() == inputBuffer->length() && bufferSize() == outputBuffer->length()
         && m_bufferReadWriteIndex + framesToProcess <= bufferSize();
     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 numberOfInputChannels = inputBus->numberOfChannels();

     bool channelsAreGood = (numberOfInputChannels == 1 || numberOfInputChannels == 2) && outputBus->numberOfChannels() == 2;
     ASSERT(channelsAreGood);
     if (!channelsAreGood)
         return;

     float* sourceL = inputBus->channel(0)->data();
     float* sourceR = numberOfInputChannels > 1 ? inputBus->channel(1)->data() : 0;
     float* destinationL = outputBus->channel(0)->data();
     float* destinationR = outputBus->channel(1)->data();

     // Copy from the input to the input buffer.  See "buffersAreGood" check above for safety.
     size_t bytesToCopy = sizeof(float) * framesToProcess;
     memcpy(inputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);

     if (numberOfInputChannels == 2)
         memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceR, bytesToCopy);
     else if (numberOfInputChannels == 1) {
         // If the input is mono, then also copy the mono input to the right channel of the AudioBuffer which the AudioProcessingEvent uses.
         // FIXME: it is likely the audio API will evolve to present an AudioBuffer with the same number of channels as our input.
         memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
     }

     // Copy from the output buffer to the output.  See "buffersAreGood" check above for safety.
     memcpy(destinationL, outputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, bytesToCopy);
     memcpy(destinationR, outputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, bytesToCopy);

     // 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(fireProcessEventDispatch, this);
         }

         swapBuffers();
     }
 }

 void JavaScriptAudioNode::fireProcessEventDispatch(void* userData)
 {
     JavaScriptAudioNode* jsAudioNode = static_cast<JavaScriptAudioNode*>(userData);
     ASSERT(jsAudioNode);
     if (!jsAudioNode)
         return;

     jsAudioNode->fireProcessEvent();

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

 void JavaScriptAudioNode::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(inputBuffer && outputBuffer);
     if (!inputBuffer || !outputBuffer)
         return;

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

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

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

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

 ScriptExecutionContext* JavaScriptAudioNode::scriptExecutionContext() const
 {
     return const_cast<JavaScriptAudioNode*>(this)->context()->document();
 }

 } // 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 "JavaScriptAudioNode.h"

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

	namespace WebCore {

	const size_t DefaultBufferSize = 4096;

	PassRefPtr<JavaScriptAudioNode> JavaScriptAudioNode::create(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
	{
	return adoptRef(new JavaScriptAudioNode(context, sampleRate, bufferSize, numberOfInputs, numberOfOutputs));
	}

	JavaScriptAudioNode::JavaScriptAudioNode(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
	: AudioNode(context, sampleRate)
	, m_doubleBufferIndex(0)
	, m_doubleBufferIndexForEvent(0)
	, m_bufferSize(bufferSize)
	, m_bufferReadWriteIndex(0)
	, m_isRequestOutstanding(false)
	{
	// Check for valid buffer size.
	switch (bufferSize) {
	case 256:
	case 512:
	case 1024:
	case 2048:
	case 4096:
	case 8192:
	case 16384:
	m_bufferSize = bufferSize;
	break;
	default:
	m_bufferSize = DefaultBufferSize;
	}

	// Regardless of the allowed buffer sizes above, we still need to process at the granularity of the AudioNode.
	if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
	m_bufferSize = AudioNode::ProcessingSizeInFrames;

	// FIXME: Right now we're hardcoded to single input and single output.
	// Although the specification says this is OK for a simple implementation, multiple inputs and outputs would be good.
	ASSERT_UNUSED(numberOfInputs, numberOfInputs == 1);
	ASSERT_UNUSED(numberOfOutputs, numberOfOutputs == 1);
	addInput(adoptPtr(new AudioNodeInput(this)));
	addOutput(adoptPtr(new AudioNodeOutput(this, 2)));

	setType(NodeTypeJavaScript);

	initialize();
	}

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

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

	double 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) {
	m_inputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
	m_outputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
	}

	AudioNode::initialize();
	}

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

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

	AudioNode::uninitialize();
	}

	JavaScriptAudioNode* JavaScriptAudioNode::toJavaScriptAudioNode()
	{
	return this;
	}

	void JavaScriptAudioNode::process(size_t framesToProcess)
	{
	// Discussion about inputs and outputs:
	// As in other AudioNodes, JavaScriptAudioNode 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.
	bool buffersAreGood = inputBuffer && outputBuffer && bufferSize() == inputBuffer->length() && bufferSize() == outputBuffer->length()
	&& m_bufferReadWriteIndex + framesToProcess <= bufferSize();
	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 numberOfInputChannels = inputBus->numberOfChannels();

	bool channelsAreGood = (numberOfInputChannels == 1 \|\| numberOfInputChannels == 2) && outputBus->numberOfChannels() == 2;
	ASSERT(channelsAreGood);
	if (!channelsAreGood)
	return;

	float* sourceL = inputBus->channel(0)->data();
	float* sourceR = numberOfInputChannels > 1 ? inputBus->channel(1)->data() : 0;
	float* destinationL = outputBus->channel(0)->data();
	float* destinationR = outputBus->channel(1)->data();

	// Copy from the input to the input buffer. See "buffersAreGood" check above for safety.
	size_t bytesToCopy = sizeof(float) * framesToProcess;
	memcpy(inputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);

	if (numberOfInputChannels == 2)
	memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceR, bytesToCopy);
	else if (numberOfInputChannels == 1) {
	// If the input is mono, then also copy the mono input to the right channel of the AudioBuffer which the AudioProcessingEvent uses.
	// FIXME: it is likely the audio API will evolve to present an AudioBuffer with the same number of channels as our input.
	memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
	}

	// Copy from the output buffer to the output. See "buffersAreGood" check above for safety.
	memcpy(destinationL, outputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, bytesToCopy);
	memcpy(destinationR, outputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, bytesToCopy);

	// 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(fireProcessEventDispatch, this);
	}

	swapBuffers();
	}
	}

	void JavaScriptAudioNode::fireProcessEventDispatch(void* userData)
	{
	JavaScriptAudioNode* jsAudioNode = static_cast<JavaScriptAudioNode*>(userData);
	ASSERT(jsAudioNode);
	if (!jsAudioNode)
	return;

	jsAudioNode->fireProcessEvent();

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

	void JavaScriptAudioNode::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(inputBuffer && outputBuffer);
	if (!inputBuffer \|\| !outputBuffer)
	return;

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

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

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

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

	ScriptExecutionContext* JavaScriptAudioNode::scriptExecutionContext() const
	{
	return const_cast<JavaScriptAudioNode*>(this)->context()->document();
	}

	} // namespace WebCore

	#endif // ENABLE(WEB_AUDIO)