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/*
* 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 "AudioUtilities.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, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
{
auto node = adoptRef(*new ScriptProcessorNode(context, bufferSize, numberOfInputChannels, numberOfOutputChannels));
node->suspendIfNeeded();
return node;
}
ScriptProcessorNode::ScriptProcessorNode(BaseAudioContext& context, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
: AudioNode(context, NodeTypeJavaScript)
, ActiveDOMObject(context.scriptExecutionContext())
, m_bufferSize(bufferSize)
, m_numberOfInputChannels(numberOfInputChannels)
, m_numberOfOutputChannels(numberOfOutputChannels)
, m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioUtilities::renderQuantumSize, false))
{
// Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode.
if (m_bufferSize < AudioUtilities::renderQuantumSize)
m_bufferSize = AudioUtilities::renderQuantumSize;
ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels);
initializeDefaultNodeOptions(numberOfInputChannels, ChannelCountMode::Explicit, ChannelInterpretation::Speakers);
addInput();
addOutput(numberOfOutputChannels);
initialize();
}
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 < bufferCount; ++i) {
// We prevent detaching the AudioBuffers here since we pass those to JS and reuse them.
m_inputBuffers[i] = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate, AudioBuffer::LegacyPreventDetaching::Yes) : nullptr;
m_outputBuffers[i] = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate, AudioBuffer::LegacyPreventDetaching::Yes) : nullptr;
}
AudioNode::initialize();
}
RefPtr<AudioBuffer> ScriptProcessorNode::createInputBufferForJS(AudioBuffer* inputBuffer) const
{
if (!inputBuffer)
return nullptr;
// As an optimization, we reuse the same buffer as last time when possible.
if (!m_cachedInputBufferForJS || !inputBuffer->copyTo(*m_cachedInputBufferForJS))
m_cachedInputBufferForJS = inputBuffer->clone();
return m_cachedInputBufferForJS;
}
RefPtr<AudioBuffer> ScriptProcessorNode::createOutputBufferForJS(AudioBuffer& outputBuffer) const
{
// As an optimization, we reuse the same buffer as last time when possible.
if (!m_cachedOutputBufferForJS || !m_cachedOutputBufferForJS->topologyMatches(outputBuffer))
m_cachedOutputBufferForJS = outputBuffer.clone(AudioBuffer::ShouldCopyChannelData::No);
else
m_cachedOutputBufferForJS->zero();
return m_cachedOutputBufferForJS;
}
void ScriptProcessorNode::uninitialize()
{
if (!isInitialized())
return;
for (unsigned i = 0; i < bufferCount; ++i) {
Locker locker { m_bufferLocks[i] };
m_inputBuffers[i] = nullptr;
m_outputBuffers[i] = nullptr;
}
AudioNode::uninitialize();
}
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 (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. The JavaScript gets its own copy
// of the buffers for safety reasons.
// 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 bufferIndex = this->bufferIndex();
ASSERT(bufferIndex < bufferCount);
if (!m_bufferLocks[bufferIndex].tryLock()) {
// 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.
outputBus->zero();
return;
}
Locker locker { AdoptLock, m_bufferLocks[bufferIndex] };
AudioBuffer* inputBuffer = m_inputBuffers[bufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[bufferIndex].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->rawChannelData(i) + 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->rawChannelData(i) + 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) {
// Heap allocations are forbidden on the audio thread for performance reasons so we need to
// explicitly allow the following allocation(s).
DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
// Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
// We only wait for script code execution when the context is an offline one for performance reasons.
if (context().isOfflineContext()) {
callOnMainThreadAndWait([this, bufferIndex, protector = Ref { *this }] {
fireProcessEvent(bufferIndex);
});
} else {
callOnMainThread([this, bufferIndex, protector = Ref { *this }] {
Locker locker { m_bufferLocks[bufferIndex] };
fireProcessEvent(bufferIndex);
});
}
swapBuffers();
}
}
void ScriptProcessorNode::fireProcessEvent(unsigned bufferIndex)
{
ASSERT(isMainThread());
AudioBuffer* inputBuffer = m_inputBuffers[bufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[bufferIndex].get();
ASSERT(outputBuffer);
if (!outputBuffer)
return;
// Avoid firing the event if the document has already gone away.
if (context().isStopped())
return;
// 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());
auto inputBufferForJS = createInputBufferForJS(inputBuffer);
auto outputBufferForJS = createOutputBufferForJS(*outputBuffer);
// Call the JavaScript event handler which will do the audio processing.
dispatchEvent(AudioProcessingEvent::create(inputBufferForJS.get(), outputBufferForJS.get(), playbackTime));
if (!outputBufferForJS->copyTo(*outputBuffer))
outputBuffer->zero();
}
ExceptionOr<void> ScriptProcessorNode::setChannelCount(unsigned channelCount)
{
ASSERT(isMainThread());
if (channelCount != this->channelCount())
return Exception { IndexSizeError, "ScriptProcessorNode's channelCount cannot be changed"_s };
return { };
}
ExceptionOr<void> ScriptProcessorNode::setChannelCountMode(ChannelCountMode mode)
{
ASSERT(isMainThread());
if (mode != this->channelCountMode())
return Exception { NotSupportedError, "ScriptProcessorNode's channelCountMode cannot be changed from 'explicit'"_s };
return { };
}
double ScriptProcessorNode::tailTime() const
{
return std::numeric_limits<double>::infinity();
}
double ScriptProcessorNode::latencyTime() const
{
return std::numeric_limits<double>::infinity();
}
bool ScriptProcessorNode::requiresTailProcessing() const
{
// Always return true since the tail and latency are never zero.
return true;
}
void ScriptProcessorNode::eventListenersDidChange()
{
m_hasAudioProcessEventListener = hasEventListeners(eventNames().audioprocessEvent);
}
bool ScriptProcessorNode::virtualHasPendingActivity() const
{
if (context().isClosed())
return false;
return m_hasAudioProcessEventListener;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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