RealtimeIncomingAudioSourceCocoa.cpp   [plain text]

/*
 * Copyright (C) 2017 Apple Inc. All rights reserved.
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 * modification, are permitted provided that the following conditions
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 * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
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#include "config.h"
#include "RealtimeIncomingAudioSourceCocoa.h"

#if USE(LIBWEBRTC)

#include "AudioStreamDescription.h"
#include "CAAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "WebAudioBufferList.h"
#include "WebAudioSourceProviderAVFObjC.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)
{
    CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate);
    auto mediaTime = PAL::toMediaTime(startTime);
    m_numberOfFrames += numberOfFrames;

    AudioStreamBasicDescription newDescription = streamDescription(sampleRate, numberOfChannels);

    // FIXME: We should not need to do the extra memory allocation and copy.
    // Instead, we should be able to directly pass audioData pointer.
    WebAudioBufferList audioBufferList { CAAudioStreamDescription(newDescription), WTF::safeCast<uint32_t>(numberOfFrames) };
    audioBufferList.buffer(0)->mDataByteSize = numberOfChannels * numberOfFrames * bitsPerSample / 8;
    audioBufferList.buffer(0)->mNumberChannels = numberOfChannels;

    if (muted())
        memset(audioBufferList.buffer(0)->mData, 0, audioBufferList.buffer(0)->mDataByteSize);
    else
        memcpy(audioBufferList.buffer(0)->mData, audioData, audioBufferList.buffer(0)->mDataByteSize);

    audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(newDescription), numberOfFrames);
}

}

#endif // USE(LIBWEBRTC)