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/*
* 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,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
* DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
* THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "config.h"
#include "RealtimeIncomingAudioSourceCocoa.h"
#if USE(LIBWEBRTC)
#include "AudioStreamDescription.h"
#include "CAAudioStreamDescription.h"
#include "LibWebRTCAudioFormat.h"
#include "Logging.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);
#if !RELEASE_LOG_DISABLED
if (!(++m_chunksReceived % 200))
ALWAYS_LOG_IF(loggerPtr(), LOGIDENTIFIER, "chunk ", m_chunksReceived);
#endif
audioSamplesAvailable(mediaTime, audioBufferList, CAAudioStreamDescription(newDescription), numberOfFrames);
}
}
#endif // USE(LIBWEBRTC)