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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 "ConvolverNode.h"
- #include "AudioBuffer.h"
- #include "AudioContext.h"
- #include "AudioNodeInput.h"
- #include "AudioNodeOutput.h"
- #include "Reverb.h"
- #include <wtf/MainThread.h>
- // 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 WebCore {
- ConvolverNode::ConvolverNode(AudioContext* context, float sampleRate)
- : AudioNode(context, sampleRate)
- , m_normalize(true)
- {
- addInput(adoptPtr(new AudioNodeInput(this)));
- addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
- // Node-specific default mixing rules.
- m_channelCount = 2;
- m_channelCountMode = ClampedMax;
- m_channelInterpretation = AudioBus::Speakers;
- setNodeType(NodeTypeConvolver);
-
- initialize();
- }
- ConvolverNode::~ConvolverNode()
- {
- uninitialize();
- }
- void ConvolverNode::process(size_t framesToProcess)
- {
- AudioBus* outputBus = output(0)->bus();
- ASSERT(outputBus);
- // Synchronize with possible dynamic changes to the impulse response.
- MutexTryLocker tryLocker(m_processLock);
- if (tryLocker.locked()) {
- if (!isInitialized() || !m_reverb.get())
- 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 ConvolverNode::reset()
- {
- MutexLocker locker(m_processLock);
- if (m_reverb.get())
- m_reverb->reset();
- }
- void ConvolverNode::initialize()
- {
- if (isInitialized())
- return;
-
- AudioNode::initialize();
- }
- void ConvolverNode::uninitialize()
- {
- if (!isInitialized())
- return;
- m_reverb.clear();
- AudioNode::uninitialize();
- }
- void ConvolverNode::setBuffer(AudioBuffer* buffer)
- {
- ASSERT(isMainThread());
-
- if (!buffer)
- return;
- unsigned numberOfChannels = buffer->numberOfChannels();
- size_t bufferLength = buffer->length();
- // The current implementation supports up to four channel impulse responses, which are interpreted as true-stereo (see Reverb class).
- bool isBufferGood = numberOfChannels > 0 && numberOfChannels <= 4 && bufferLength;
- ASSERT(isBufferGood);
- if (!isBufferGood)
- 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.
- bool useBackgroundThreads = !context()->isOfflineContext();
- OwnPtr<Reverb> reverb = adoptPtr(new Reverb(bufferBus.get(), AudioNode::ProcessingSizeInFrames, MaxFFTSize, 2, useBackgroundThreads, m_normalize));
- {
- // Synchronize with process().
- MutexLocker locker(m_processLock);
- m_reverb = reverb.release();
- m_buffer = buffer;
- }
- }
- AudioBuffer* ConvolverNode::buffer()
- {
- ASSERT(isMainThread());
- return m_buffer.get();
- }
- double ConvolverNode::tailTime() const
- {
- return m_reverb ? m_reverb->impulseResponseLength() / static_cast<double>(sampleRate()) : 0;
- }
- double ConvolverNode::latencyTime() const
- {
- return m_reverb ? m_reverb->latencyFrames() / static_cast<double>(sampleRate()) : 0;
- }
- } // namespace WebCore
- #endif // ENABLE(WEB_AUDIO)
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