Source/modules/webaudio/GainNode.cpp

/*
 * Copyright (C) 2010, Google Inc. All rights reserved.
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#include "config.h"
#if ENABLE(WEB_AUDIO)
#include "modules/webaudio/GainNode.h"

#include "modules/webaudio/AudioNodeInput.h"
#include "modules/webaudio/AudioNodeOutput.h"
#include "platform/audio/AudioBus.h"

namespace blink {

GainHandler::GainHandler(AudioNode& node, float sampleRate, AudioParamHandler& gain)
    : AudioHandler(NodeTypeGain, node, sampleRate)
    , m_lastGain(1.0)
    , m_gain(gain)
    , m_sampleAccurateGainValues(ProcessingSizeInFrames) // FIXME: can probably share temp buffer in context
{
    addInput();
    addOutput(1);

    initialize();
}

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

void GainHandler::process(size_t framesToProcess)
{
    // FIXME: for some cases there is a nice optimization to avoid processing here, and let the gain change
    // happen in the summing junction input of the AudioNode we're connected to.
    // Then we can avoid all of the following:

    AudioBus* outputBus = output(0).bus();
    ASSERT(outputBus);

    if (!isInitialized() || !input(0).isConnected()) {
        outputBus->zero();
    } else {
        AudioBus* inputBus = input(0).bus();

        if (m_gain->hasSampleAccurateValues()) {
            // Apply sample-accurate gain scaling for precise envelopes, grain windows, etc.
            ASSERT(framesToProcess <= m_sampleAccurateGainValues.size());
            if (framesToProcess <= m_sampleAccurateGainValues.size()) {
                float* gainValues = m_sampleAccurateGainValues.data();
                m_gain->calculateSampleAccurateValues(gainValues, framesToProcess);
                outputBus->copyWithSampleAccurateGainValuesFrom(*inputBus, gainValues, framesToProcess);
            }
        } else {
            // Apply the gain with de-zippering into the output bus.
            outputBus->copyWithGainFrom(*inputBus, &m_lastGain, m_gain->value());
        }
    }
}

// FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInput
//
// As soon as we know the channel count of our input, we can lazily initialize.
// Sometimes this may be called more than once with different channel counts, in which case we must safely
// uninitialize and then re-initialize with the new channel count.
void GainHandler::checkNumberOfChannelsForInput(AudioNodeInput* input)
{
    ASSERT(context()->isAudioThread());
    ASSERT(context()->isGraphOwner());

    ASSERT(input);
    ASSERT(input == &this->input(0));
    if (input != &this->input(0))
        return;

    unsigned numberOfChannels = input->numberOfChannels();

    if (isInitialized() && numberOfChannels != output(0).numberOfChannels()) {
        // We're already initialized but the channel count has changed.
        uninitialize();
    }

    if (!isInitialized()) {
        // This will propagate the channel count to any nodes connected further downstream in the graph.
        output(0).setNumberOfChannels(numberOfChannels);
        initialize();
    }

    AudioHandler::checkNumberOfChannelsForInput(input);
}

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

GainNode::GainNode(AbstractAudioContext& context, float sampleRate)
    : AudioNode(context)
    , m_gain(AudioParam::create(context, 1.0))
{
    setHandler(GainHandler::create(*this, sampleRate, m_gain->handler()));
}

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

AudioParam* GainNode::gain() const
{
    return m_gain;
}

DEFINE_TRACE(GainNode)
{
    visitor->trace(m_gain);
    AudioNode::trace(visitor);
}

} // namespace blink

#endif // ENABLE(WEB_AUDIO)