RealtimeOutgoingAudioSourceCocoa.cpp   [plain text]

/*
 * Copyright (C) 2017 Apple Inc.
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 * modification, are permitted, provided that the following conditions
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#include "config.h"
#include "RealtimeOutgoingAudioSourceCocoa.h"

#if USE(LIBWEBRTC)

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

namespace WebCore {

static inline AudioStreamBasicDescription libwebrtcAudioFormat(Float64 sampleRate, size_t channelCount)
{
    // FIXME: Microphones can have more than two channels. In such case, we should do the mix down based on AudioChannelLayoutTag.
    size_t libWebRTCChannelCount = channelCount >= 2 ? 2 : channelCount;
    AudioStreamBasicDescription streamFormat;
    FillOutASBDForLPCM(streamFormat, sampleRate, libWebRTCChannelCount, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::sampleSize, LibWebRTCAudioFormat::isFloat, LibWebRTCAudioFormat::isBigEndian, LibWebRTCAudioFormat::isNonInterleaved);
    return streamFormat;
}

RealtimeOutgoingAudioSourceCocoa::RealtimeOutgoingAudioSourceCocoa(Ref<MediaStreamTrackPrivate>&& audioSource)
    : RealtimeOutgoingAudioSource(WTFMove(audioSource))
    , m_sampleConverter(AudioSampleDataSource::create(LibWebRTCAudioFormat::sampleRate * 2))
{
}

Ref<RealtimeOutgoingAudioSource> RealtimeOutgoingAudioSource::create(Ref<MediaStreamTrackPrivate>&& audioSource)
{
    return RealtimeOutgoingAudioSourceCocoa::create(WTFMove(audioSource));
}

bool RealtimeOutgoingAudioSourceCocoa::isReachingBufferedAudioDataHighLimit()
{
    auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
    auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();

    ASSERT(writtenAudioDuration >= readAudioDuration);
    return writtenAudioDuration > readAudioDuration + 0.5;
}

bool RealtimeOutgoingAudioSourceCocoa::isReachingBufferedAudioDataLowLimit()
{
    auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
    auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();

    ASSERT(writtenAudioDuration >= readAudioDuration);
    return writtenAudioDuration < readAudioDuration + 0.1;
}

bool RealtimeOutgoingAudioSourceCocoa::hasBufferedEnoughData()
{
    auto writtenAudioDuration = m_writeCount / m_inputStreamDescription.sampleRate();
    auto readAudioDuration = m_readCount / m_outputStreamDescription.sampleRate();

    ASSERT(writtenAudioDuration >= readAudioDuration);
    return writtenAudioDuration >= readAudioDuration + 0.01;
}

void RealtimeOutgoingAudioSourceCocoa::audioSamplesAvailable(const MediaTime&, const PlatformAudioData& audioData, const AudioStreamDescription& streamDescription, size_t sampleCount)
{
    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);
    }

    // Let's skip pushing samples if we are too slow pulling them.
    if (m_skippingAudioData) {
        if (!isReachingBufferedAudioDataLowLimit())
            return;
        m_skippingAudioData = false;
    } else if (isReachingBufferedAudioDataHighLimit()) {
        m_skippingAudioData = true;
        return;
    }

    // If we change the audio track or its sample rate changes, the timestamp based on m_writeCount may be wrong.
    // FIXME: We should update m_writeCount to be valid according the new sampleRate.
    m_sampleConverter->pushSamples(MediaTime(m_writeCount, static_cast<uint32_t>(m_inputStreamDescription.sampleRate())), audioData, sampleCount);
    m_writeCount += sampleCount;

    if (!hasBufferedEnoughData())
        return;

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

void RealtimeOutgoingAudioSourceCocoa::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();

    if (isSilenced() !=  m_sampleConverter->muted())
        m_sampleConverter->setMuted(isSilenced());

    m_sampleConverter->pullAvalaibleSamplesAsChunks(bufferList, chunkSampleCount, m_readCount, [this, chunkSampleCount] {
        m_readCount += 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)