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

/*
 * Copyright (C) 2017 Apple Inc.
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 * modification, are permitted, provided that the following conditions
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 *     from this software without specific prior written permission.
 *
 * 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
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#include "config.h"
#include "RealtimeOutgoingAudioSource.h"

#if USE(LIBWEBRTC)

#include "CAAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "LibWebRTCProvider.h"

namespace WebCore {

static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
{
    AudioStreamBasicDescription streamFormat;
    FillOutASBDForLPCM(streamFormat, sampleRate, channelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
    return streamFormat;
}

RealtimeOutgoingAudioSource::RealtimeOutgoingAudioSource(Ref<RealtimeMediaSource>&& audioSource)
    : m_audioSource(WTFMove(audioSource))
    , m_sampleConverter(AudioSampleDataSource::create(LibWebRTCAudioFormat::sampleRate * 2))
{
    m_audioSource->addObserver(*this);
}

void RealtimeOutgoingAudioSource::stop()
{
    m_audioSource->removeObserver(*this);
}

void RealtimeOutgoingAudioSource::sourceMutedChanged()
{
    m_muted = m_audioSource->muted();
    m_sampleConverter->setMuted(m_muted || !m_enabled);
}

void RealtimeOutgoingAudioSource::sourceEnabledChanged()
{
    m_enabled = m_audioSource->enabled();
    m_sampleConverter->setMuted(m_muted || !m_enabled);
}

void RealtimeOutgoingAudioSource::audioSamplesAvailable(const MediaTime& time, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
{
    ASSERT(streamDescription.numberOfChannels() <= 2);

    if (m_inputStreamDescription != streamDescription) {
        m_inputStreamDescription = toCAAudioStreamDescription(streamDescription);
        auto status  = m_sampleConverter->setInputFormat(m_inputStreamDescription);
        ASSERT_UNUSED(status, !status);

        m_outputStreamDescription = libwebrtcAudioFormat(LibWebRTCAudioFormat::sampleRate, streamDescription.numberOfChannels());
        status = m_sampleConverter->setOutputFormat(m_outputStreamDescription.streamDescription());
        ASSERT(!status);
    }
    m_sampleConverter->pushSamples(time, audioData, sampleCount);

    LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this)] {
        protectedThis->pullAudioData();
    });
}

void RealtimeOutgoingAudioSource::pullAudioData()
{
    // libwebrtc expects 10 ms chunks.
    size_t chunkSampleCount = m_outputStreamDescription.sampleRate() / 100;
    size_t bufferSize = chunkSampleCount * LibWebRTCAudioFormat::sampleByteSize * m_outputStreamDescription.numberOfChannels();
    m_audioBuffer.grow(bufferSize);

    AudioBufferList bufferList;
    bufferList.mNumberBuffers = 1;
    bufferList.mBuffers[0].mNumberChannels = m_outputStreamDescription.numberOfChannels();
    bufferList.mBuffers[0].mDataByteSize = bufferSize;
    bufferList.mBuffers[0].mData = m_audioBuffer.data();

    m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_startFrame, [this, chunkSampleCount] {
        m_startFrame += chunkSampleCount;
        for (auto sink : m_sinks)
            sink->OnData(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, m_outputStreamDescription.sampleRate(), m_outputStreamDescription.numberOfChannels(), chunkSampleCount);
    });
}

} // namespace WebCore

#endif // USE(LIBWEBRTC)