Source/WebCore/platform/mediastream/mac/AudioMediaStreamTrackRendererUnit.cpp

/*
 * Copyright (C) 2020 Apple 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
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#include "config.h"
#include "AudioMediaStreamTrackRendererUnit.h"

#if ENABLE(MEDIA_STREAM)

#include "AudioSampleDataSource.h"
#include "AudioSession.h"
#include "CAAudioStreamDescription.h"
#include "Logging.h"

#include <pal/spi/cocoa/AudioToolboxSPI.h>
#include <wtf/Lock.h>

#include <pal/cf/CoreMediaSoftLink.h>

namespace WebCore {

AudioMediaStreamTrackRendererUnit& AudioMediaStreamTrackRendererUnit::singleton()
{
    static NeverDestroyed<AudioMediaStreamTrackRendererUnit> registry;
    return registry;
}

AudioMediaStreamTrackRendererUnit::~AudioMediaStreamTrackRendererUnit()
{
    stop();
}

void AudioMediaStreamTrackRendererUnit::addSource(Ref<AudioSampleDataSource>&& source)
{
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererUnit::addSource");

    {
        auto locker = holdLock(m_sourcesLock);
        ASSERT(!m_sources.contains(source.get()));
        m_sources.add(WTFMove(source));
        m_sourcesCopy = copyToVector(m_sources);
        m_shouldUpdateRenderSources = true;
    }
    start();
}

void AudioMediaStreamTrackRendererUnit::removeSource(AudioSampleDataSource& source)
{
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererUnit::removeSource");

    bool shouldStop = false;
    {
        auto locker = holdLock(m_sourcesLock);
        m_sources.remove(source);
        shouldStop = m_sources.isEmpty();
        m_sourcesCopy = copyToVector(m_sources);
        m_shouldUpdateRenderSources = true;
    }
    if (shouldStop)
        stop();
}

void AudioMediaStreamTrackRendererUnit::start()
{
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererUnit::start");

    if (m_isStarted)
        return;

    createAudioUnitIfNeeded();
    if (!m_remoteIOUnit)
        return;

    if (auto error = AudioOutputUnitStart(m_remoteIOUnit)) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::start AudioOutputUnitStart failed, error = %d", error);
        AudioComponentInstanceDispose(m_remoteIOUnit);
        m_remoteIOUnit = nullptr;
    }
    m_isStarted = true;
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererUnit is started");
}

void AudioMediaStreamTrackRendererUnit::stop()
{
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererUnit::stop");

    if (!m_remoteIOUnit)
        return;

    if (m_isStarted) {
        AudioOutputUnitStop(m_remoteIOUnit);
        m_isStarted = false;
    }

    AudioComponentInstanceDispose(m_remoteIOUnit);
    m_remoteIOUnit = nullptr;
}

CAAudioStreamDescription* AudioMediaStreamTrackRendererUnit::formatDescription()
{
    createAudioUnitIfNeeded();
    return m_outputDescription.get();
}

void AudioMediaStreamTrackRendererUnit::createAudioUnitIfNeeded()
{
    ASSERT(!m_remoteIOUnit || m_outputDescription);
    if (m_remoteIOUnit)
        return;

    CAAudioStreamDescription outputDescription;
    AudioComponentInstance remoteIOUnit { nullptr };

    AudioComponentDescription ioUnitDescription { kAudioUnitType_Output, 0, kAudioUnitManufacturer_Apple, 0, 0 };
#if PLATFORM(IOS_FAMILY)
    ioUnitDescription.componentSubType = kAudioUnitSubType_RemoteIO;
#else
    ioUnitDescription.componentSubType = kAudioUnitSubType_DefaultOutput;
#endif

    AudioComponent ioComponent = AudioComponentFindNext(nullptr, &ioUnitDescription);
    ASSERT(ioComponent);
    if (!ioComponent) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to find remote IO unit component");
        return;
    }

    auto error = AudioComponentInstanceNew(ioComponent, &remoteIOUnit);
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to open vpio unit, error = %d", error);
        return;
    }

#if PLATFORM(IOS_FAMILY)
    UInt32 param = 1;
    error = AudioUnitSetProperty(remoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(param));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to enable vpio unit output, error = %d", error);
        return;
    }
#endif

    AURenderCallbackStruct callback = { inputProc, nullptr };
    error = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callback, sizeof(callback));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to set vpio unit speaker proc, error = %d", error);
        return;
    }

    UInt32 size = sizeof(outputDescription.streamDescription());
    error  = AudioUnitGetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), &size);
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to get input stream format, error = %d", error);
        return;
    }

    outputDescription.streamDescription().mSampleRate = AudioSession::sharedSession().sampleRate();

    error = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), sizeof(outputDescription.streamDescription()));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit unable to set input stream format, error = %d", error);
        return;
    }

    error = AudioUnitInitialize(remoteIOUnit);
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererUnit::createAudioUnit AudioUnitInitialize() failed, error = %d", error);
        return;
    }

    m_outputDescription = makeUnique<CAAudioStreamDescription>(outputDescription);
    m_remoteIOUnit = remoteIOUnit;
}

OSStatus AudioMediaStreamTrackRendererUnit::render(UInt32 sampleCount, AudioBufferList& ioData, UInt32 /*inBusNumber*/, const AudioTimeStamp& timeStamp, AudioUnitRenderActionFlags& actionFlags)
{
    ASSERT(!isMainThread());
    if (m_shouldUpdateRenderSources) {
        auto locker = tryHoldLock(m_sourcesLock);
        if (!locker)
            return 0;

        m_renderSources = WTFMove(m_sourcesCopy);
        m_shouldUpdateRenderSources = false;
    }

    if (m_renderSources.isEmpty()) {
        if (m_outputDescription)
            AudioSampleBufferList::zeroABL(ioData, static_cast<size_t>(sampleCount * m_outputDescription->bytesPerFrame()));
        actionFlags = kAudioUnitRenderAction_OutputIsSilence;
        return 0;
    }

    // Mix all sources.
    bool isFirstSource = true;
    for (auto& source : m_renderSources) {
        source->pullSamples(ioData, static_cast<size_t>(sampleCount), timeStamp.mSampleTime, timeStamp.mHostTime, isFirstSource ? AudioSampleDataSource::Copy : AudioSampleDataSource::Mix);
        isFirstSource = false;
    }
    return 0;
}

OSStatus AudioMediaStreamTrackRendererUnit::inputProc(void*, AudioUnitRenderActionFlags* actionFlags, const AudioTimeStamp* timeStamp, UInt32 inBusNumber, UInt32 sampleCount, AudioBufferList* ioData)
{
    return AudioMediaStreamTrackRendererUnit::singleton().render(sampleCount, *ioData, inBusNumber, *timeStamp, *actionFlags);
}

} // namespace WebCore

#endif // ENABLE(MEDIA_STREAM)