/* * 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 "modules/webaudio/ScriptProcessorNode.h" #include "bindings/core/v8/ExceptionState.h" #include "core/dom/CrossThreadTask.h" #include "core/dom/ExceptionCode.h" #include "core/dom/ExecutionContext.h" #include "modules/webaudio/AbstractAudioContext.h" #include "modules/webaudio/AudioBuffer.h" #include "modules/webaudio/AudioNodeInput.h" #include "modules/webaudio/AudioNodeOutput.h" #include "modules/webaudio/AudioProcessingEvent.h" #include "public/platform/Platform.h" namespace blink { ScriptProcessorHandler::ScriptProcessorHandler(AudioNode& node, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) : AudioHandler(NodeTypeJavaScript, node, sampleRate) , m_doubleBufferIndex(0) , m_bufferSize(bufferSize) , m_bufferReadWriteIndex(0) , m_numberOfInputChannels(numberOfInputChannels) , m_numberOfOutputChannels(numberOfOutputChannels) , m_internalInputBus(AudioBus::create(numberOfInputChannels, ProcessingSizeInFrames, false)) { // Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode. if (m_bufferSize < ProcessingSizeInFrames) m_bufferSize = ProcessingSizeInFrames; ASSERT(numberOfInputChannels <= AbstractAudioContext::maxNumberOfChannels()); addInput(); addOutput(numberOfOutputChannels); m_channelCount = numberOfInputChannels; m_channelCountMode = Explicit; initialize(); } PassRefPtr ScriptProcessorHandler::create(AudioNode& node, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) { return adoptRef(new ScriptProcessorHandler(node, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels)); } ScriptProcessorHandler::~ScriptProcessorHandler() { uninitialize(); } void ScriptProcessorHandler::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) { AudioBuffer* inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0; AudioBuffer* outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0; m_inputBuffers.append(inputBuffer); m_outputBuffers.append(outputBuffer); } AudioHandler::initialize(); } void ScriptProcessorHandler::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 = input(0).bus(); AudioBus* outputBus = 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->getChannelData(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->getChannelData(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. // The audio thread can't block on this lock, so we call tryLock() instead. MutexTryLocker tryLocker(m_processEventLock); if (!tryLocker.locked()) { // 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 if (context()->executionContext()) { // Fire the event on the main thread with the appropriate buffer // index. context()->executionContext()->postTask(BLINK_FROM_HERE, createCrossThreadTask(&ScriptProcessorHandler::fireProcessEvent, this, m_doubleBufferIndex)); } swapBuffers(); } } void ScriptProcessorHandler::fireProcessEvent(unsigned doubleBufferIndex) { ASSERT(isMainThread()); ASSERT(doubleBufferIndex < 2); if (doubleBufferIndex > 1) return; AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get(); AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get(); ASSERT(outputBuffer); if (!outputBuffer) return; // Avoid firing the event if the document has already gone away. if (node() && context() && context()->executionContext()) { // This synchronizes with process(). MutexLocker processLocker(m_processEventLock); // Calculate a playbackTime with the buffersize which needs to be processed each time onaudioprocess is called. // The outputBuffer being passed to JS will be played after exhuasting previous outputBuffer by double-buffering. double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast(context()->sampleRate()); // Call the JavaScript event handler which will do the audio processing. node()->dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime)); } } double ScriptProcessorHandler::tailTime() const { return std::numeric_limits::infinity(); } double ScriptProcessorHandler::latencyTime() const { return std::numeric_limits::infinity(); } void ScriptProcessorHandler::setChannelCount(unsigned long channelCount, ExceptionState& exceptionState) { ASSERT(isMainThread()); AbstractAudioContext::AutoLocker locker(context()); if (channelCount != m_channelCount) { exceptionState.throwDOMException( NotSupportedError, "channelCount cannot be changed from " + String::number(m_channelCount) + " to " + String::number(channelCount)); } } void ScriptProcessorHandler::setChannelCountMode(const String& mode, ExceptionState& exceptionState) { ASSERT(isMainThread()); AbstractAudioContext::AutoLocker locker(context()); if ((mode == "max") || (mode == "clamped-max")) { exceptionState.throwDOMException( NotSupportedError, "channelCountMode cannot be changed from 'explicit' to '" + mode + "'"); } } // ---------------------------------------------------------------- ScriptProcessorNode::ScriptProcessorNode(AbstractAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) : AudioNode(context) { setHandler(ScriptProcessorHandler::create(*this, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels)); } static size_t chooseBufferSize() { // Choose a buffer size based on the audio hardware buffer size. Arbitarily make it a power of // two that is 4 times greater than the hardware buffer size. // FIXME: What is the best way to choose this? size_t hardwareBufferSize = Platform::current()->audioHardwareBufferSize(); size_t bufferSize = 1 << static_cast(log2(4 * hardwareBufferSize) + 0.5); if (bufferSize < 256) return 256; if (bufferSize > 16384) return 16384; return bufferSize; } ScriptProcessorNode* ScriptProcessorNode::create(AbstractAudioContext& context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels) { // Check for valid buffer size. switch (bufferSize) { case 0: bufferSize = chooseBufferSize(); break; case 256: case 512: case 1024: case 2048: case 4096: case 8192: case 16384: break; default: return nullptr; } if (!numberOfInputChannels && !numberOfOutputChannels) return nullptr; if (numberOfInputChannels > AbstractAudioContext::maxNumberOfChannels()) return nullptr; if (numberOfOutputChannels > AbstractAudioContext::maxNumberOfChannels()) return nullptr; return new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels); } size_t ScriptProcessorNode::bufferSize() const { return static_cast(handler()).bufferSize(); } } // namespace blink