Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp

/*
 * Copyright (C) 2012, Google Inc. All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
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 *
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 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
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#include "config.h"

#if ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM)

#include "MediaStreamAudioSourceNode.h"

#include "AudioContext.h"
#include "AudioNodeOutput.h"
#include "Logging.h"
#include <wtf/IsoMallocInlines.h>
#include <wtf/Locker.h>

namespace WebCore {

WTF_MAKE_ISO_ALLOCATED_IMPL(MediaStreamAudioSourceNode);

Ref<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext& context, MediaStream& mediaStream, MediaStreamTrack& audioTrack)
{
    return adoptRef(*new MediaStreamAudioSourceNode(context, mediaStream, audioTrack));
}

MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext& context, MediaStream& mediaStream, MediaStreamTrack& audioTrack)
    : AudioNode(context, context.sampleRate())
    , m_mediaStream(mediaStream)
    , m_audioTrack(audioTrack)
{
    setNodeType(NodeTypeMediaStreamAudioSource);
    
    AudioSourceProvider* audioSourceProvider = m_audioTrack->audioSourceProvider();
    ASSERT(audioSourceProvider);

    audioSourceProvider->setClient(this);
    
    // Default to stereo. This could change depending on the format of the MediaStream's audio track.
    addOutput(makeUnique<AudioNodeOutput>(this, 2));

    initialize();
}

MediaStreamAudioSourceNode::~MediaStreamAudioSourceNode()
{
    AudioSourceProvider* audioSourceProvider = m_audioTrack->audioSourceProvider();
    ASSERT(audioSourceProvider);
    audioSourceProvider->setClient(nullptr);
    uninitialize();
}

void MediaStreamAudioSourceNode::setFormat(size_t numberOfChannels, float sourceSampleRate)
{
    float sampleRate = this->sampleRate();
    if (numberOfChannels == m_sourceNumberOfChannels && sourceSampleRate == m_sourceSampleRate)
        return;

    // The sample-rate must be equal to the context's sample-rate.
    if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels()) {
        // process() will generate silence for these uninitialized values.
        LOG(Media, "MediaStreamAudioSourceNode::setFormat(%u, %f) - unhandled format change", static_cast<unsigned>(numberOfChannels), sourceSampleRate);
        m_sourceNumberOfChannels = 0;
        return;
    }

    // Synchronize with process().
    std::lock_guard<Lock> lock(m_processMutex);

    m_sourceNumberOfChannels = numberOfChannels;
    m_sourceSampleRate = sourceSampleRate;

    if (sourceSampleRate == sampleRate)
        m_multiChannelResampler = nullptr;
    else {
        double scaleFactor = sourceSampleRate / sampleRate;
        m_multiChannelResampler = makeUnique<MultiChannelResampler>(scaleFactor, numberOfChannels);
    }

    m_sourceNumberOfChannels = numberOfChannels;

    {
        // The context must be locked when changing the number of output channels.
        AudioContext::AutoLocker contextLocker(context());

        // Do any necesssary re-configuration to the output's number of channels.
        output(0)->setNumberOfChannels(numberOfChannels);
    }
}

void MediaStreamAudioSourceNode::process(size_t numberOfFrames)
{
    AudioBus* outputBus = output(0)->bus();
    AudioSourceProvider* provider = m_audioTrack->audioSourceProvider();

    if (!mediaStream() || !m_sourceNumberOfChannels || !m_sourceSampleRate || !provider) {
        outputBus->zero();
        return;
    }

    // Use std::try_to_lock to avoid contention in the real-time audio thread.
    // If we fail to acquire the lock then the MediaStream must be in the middle of
    // a format change, so we output silence in this case.
    std::unique_lock<Lock> lock(m_processMutex, std::try_to_lock);
    if (!lock.owns_lock()) {
        // We failed to acquire the lock.
        outputBus->zero();
        return;
    }
    if (m_sourceNumberOfChannels != outputBus->numberOfChannels()) {
        outputBus->zero();
        return;
    }

    if (m_multiChannelResampler.get()) {
        ASSERT(m_sourceSampleRate != sampleRate());
        m_multiChannelResampler->process(provider, outputBus, numberOfFrames);
    } else {
        // Bypass the resampler completely if the source is at the context's sample-rate.
        ASSERT(m_sourceSampleRate == sampleRate());
        provider->provideInput(outputBus, numberOfFrames);
    }
}

} // namespace WebCore

#endif // ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM)