| /* |
| * 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 <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 |
| |
| CMTime startTime = CMTimeMake(m_numberOfFrames, sampleRate); |
| auto mediaTime = PAL::toMediaTime(startTime); |
| m_numberOfFrames += numberOfFrames; |
| |
| 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); |
| } |
| |
| 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) |