RealtimeIncomingAudioSourceCocoa.cpp   [plain text]

/*
 * Copyright (C) 2017 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 THE COPYRIGHT HOLDERS AND 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 THE COPYRIGHT
 * OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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 * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
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#include "config.h"
#include "RealtimeIncomingAudioSourceCocoa.h"

#if USE(LIBWEBRTC)

#include "AudioStreamDescription.h"
#include "CAAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "Logging.h"
#include <pal/avfoundation/MediaTimeAVFoundation.h>

#include <pal/cf/CoreMediaSoftLink.h>

namespace WebCore {
using namespace PAL;

Ref<RealtimeIncomingAudioSource> RealtimeIncomingAudioSource::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
{
    auto source = RealtimeIncomingAudioSourceCocoa::create(WTFMove(audioTrack), WTFMove(audioTrackId));
    source->start();
    return WTFMove(source);
}

Ref<RealtimeIncomingAudioSourceCocoa> RealtimeIncomingAudioSourceCocoa::create(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
{
    return adoptRef(*new RealtimeIncomingAudioSourceCocoa(WTFMove(audioTrack), WTFMove(audioTrackId)));
}

RealtimeIncomingAudioSourceCocoa::RealtimeIncomingAudioSourceCocoa(rtc::scoped_refptr<webrtc::AudioTrackInterface>&& audioTrack, String&& audioTrackId)
    : RealtimeIncomingAudioSource(WTFMove(audioTrack), WTFMove(audioTrackId))
{
}

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

void RealtimeIncomingAudioSourceCocoa::OnData(const void* audioData, int bitsPerSample, int sampleRate, size_t numberOfChannels, size_t numberOfFrames)
{
#if !RELEASE_LOG_DISABLED
    if (!(++m_chunksReceived % 200)) {
        callOnMainThread([identifier = LOGIDENTIFIER, this, protectedThis = makeRef(*this), chunksReceived = m_chunksReceived] {
            ALWAYS_LOG_IF(loggerPtr(), identifier, "chunk ", chunksReceived);
        });
    }
#endif

    if (!m_audioBufferList || m_sampleRate != sampleRate || m_numberOfChannels != numberOfChannels) {
        callOnMainThread([identifier = LOGIDENTIFIER, this, protectedThis = makeRef(*this), sampleRate, numberOfChannels] {
            ALWAYS_LOG_IF(loggerPtr(), identifier, "new audio buffer list for sampleRate ", sampleRate, " and ", numberOfChannels, " channel(s)");
        });

        m_sampleRate = sampleRate;
        m_numberOfChannels = numberOfChannels;
        m_streamDescription = streamDescription(sampleRate, numberOfChannels);
        m_audioBufferList = makeUnique<WebAudioBufferList>(m_streamDescription);
        if (m_sampleRate && m_numberOfFrames)
            m_numberOfFrames = m_numberOfFrames * sampleRate / m_sampleRate;
        else
            m_numberOfFrames = 0;
    }

    CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
    auto mediaTime = PAL::toMediaTime(startTime);
    m_numberOfFrames += numberOfFrames;

    auto& bufferList = *m_audioBufferList->buffer(0);
    bufferList.mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
    bufferList.mNumberChannels = numberOfChannels;
    bufferList.mData = const_cast<void*>(audioData);

    audioSamplesAvailable(mediaTime, *m_audioBufferList, m_streamDescription, numberOfFrames);
}

}

#endif // USE(LIBWEBRTC)