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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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