third_party/WebKit/Source/modules/webaudio/ConvolverNode.cpp

/*
 * 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/ConvolverNode.h"
#include "bindings/core/v8/ExceptionState.h"
#include "core/dom/ExceptionCode.h"
#include "modules/webaudio/AudioBuffer.h"
#include "modules/webaudio/AudioNodeInput.h"
#include "modules/webaudio/AudioNodeOutput.h"
#include "platform/audio/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 blink {

ConvolverHandler::ConvolverHandler(AudioNode& node, float sampleRate)
    : AudioHandler(NodeTypeConvolver, node, sampleRate)
    , m_normalize(true)
{
    addInput();
    addOutput(2);

    // Node-specific default mixing rules.
    m_channelCount = 2;
    m_channelCountMode = ClampedMax;
    m_channelInterpretation = AudioBus::Speakers;

    initialize();
}

PassRefPtr<ConvolverHandler> ConvolverHandler::create(AudioNode& node, float sampleRate)
{
    return adoptRef(new ConvolverHandler(node, sampleRate));
}

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

void ConvolverHandler::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) {
            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 ConvolverHandler::setBuffer(AudioBuffer* buffer, ExceptionState& exceptionState)
{
    ASSERT(isMainThread());

    if (!buffer)
        return;

    if (buffer->sampleRate() != context()->sampleRate()) {
        exceptionState.throwDOMException(
            NotSupportedError,
            "The buffer sample rate of " + String::number(buffer->sampleRate())
            + " does not match the context rate of " + String::number(context()->sampleRate())
            + " Hz.");
        return;
    }

    unsigned numberOfChannels = buffer->numberOfChannels();
    size_t bufferLength = buffer->length();

    // The current implementation supports only 1-, 2-, or 4-channel impulse responses, with the
    // 4-channel response being interpreted as true-stereo (see Reverb class).
    bool isChannelCountGood = numberOfChannels == 1 || numberOfChannels == 2 || numberOfChannels == 4;

    if (!isChannelCountGood) {
        exceptionState.throwDOMException(
            NotSupportedError,
            "The buffer must have 1, 2, or 4 channels, not " + String::number(numberOfChannels));
        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.
    OwnPtr<Reverb> reverb = adoptPtr(new Reverb(bufferBus.get(), ProcessingSizeInFrames, MaxFFTSize, 2, context() && context()->hasRealtimeConstraint(), m_normalize));

    {
        // Synchronize with process().
        MutexLocker locker(m_processLock);
        m_reverb = reverb.release();
        m_buffer = buffer;
    }
}

AudioBuffer* ConvolverHandler::buffer()
{
    ASSERT(isMainThread());
    return m_buffer.get();
}

double ConvolverHandler::tailTime() const
{
    MutexTryLocker tryLocker(m_processLock);
    if (tryLocker.locked())
        return m_reverb ? m_reverb->impulseResponseLength() / static_cast<double>(sampleRate()) : 0;
    // Since we don't want to block the Audio Device thread, we return a large value
    // instead of trying to acquire the lock.
    return std::numeric_limits<double>::infinity();
}

double ConvolverHandler::latencyTime() const
{
    MutexTryLocker tryLocker(m_processLock);
    if (tryLocker.locked())
        return m_reverb ? m_reverb->latencyFrames() / static_cast<double>(sampleRate()) : 0;
    // Since we don't want to block the Audio Device thread, we return a large value
    // instead of trying to acquire the lock.
    return std::numeric_limits<double>::infinity();
}

// ----------------------------------------------------------------

ConvolverNode::ConvolverNode(AbstractAudioContext& context, float sampleRate)
    : AudioNode(context)
{
    setHandler(ConvolverHandler::create(*this, sampleRate));
}

ConvolverNode* ConvolverNode::create(AbstractAudioContext& context, float sampleRate)
{
    return new ConvolverNode(context, sampleRate);
}

ConvolverHandler& ConvolverNode::convolverHandler() const
{
    return static_cast<ConvolverHandler&>(handler());
}

AudioBuffer* ConvolverNode::buffer() const
{
    return convolverHandler().buffer();
}

void ConvolverNode::setBuffer(AudioBuffer* newBuffer, ExceptionState& exceptionState)
{
    convolverHandler().setBuffer(newBuffer, exceptionState);
}

bool ConvolverNode::normalize() const
{
    return convolverHandler().normalize();
}

void ConvolverNode::setNormalize(bool normalize)
{
    convolverHandler().setNormalize(normalize);
}

} // namespace blink