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| 1 /* | |
| 2 * Copyright 2011 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/api/audiotrack.h" | |
| 12 | |
| 13 #include "webrtc/base/checks.h" | |
| 14 | |
| 15 using rtc::scoped_refptr; | |
| 16 | |
| 17 namespace webrtc { | |
| 18 | |
| 19 const char MediaStreamTrackInterface::kAudioKind[] = "audio"; | |
| 20 | |
| 21 // static | |
| 22 scoped_refptr<AudioTrack> AudioTrack::Create( | |
| 23 const std::string& id, | |
| 24 const scoped_refptr<AudioSourceInterface>& source) { | |
| 25 return new rtc::RefCountedObject<AudioTrack>(id, source); | |
| 26 } | |
| 27 | |
| 28 AudioTrack::AudioTrack(const std::string& label, | |
| 29 const scoped_refptr<AudioSourceInterface>& source) | |
| 30 : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) { | |
| 31 if (audio_source_) { | |
| 32 audio_source_->RegisterObserver(this); | |
| 33 OnChanged(); | |
| 34 } | |
| 35 } | |
| 36 | |
| 37 AudioTrack::~AudioTrack() { | |
| 38 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 39 set_state(MediaStreamTrackInterface::kEnded); | |
| 40 if (audio_source_) | |
| 41 audio_source_->UnregisterObserver(this); | |
| 42 } | |
| 43 | |
| 44 std::string AudioTrack::kind() const { | |
| 45 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 46 return kAudioKind; | |
| 47 } | |
| 48 | |
| 49 AudioSourceInterface* AudioTrack::GetSource() const { | |
| 50 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 51 return audio_source_.get(); | |
| 52 } | |
| 53 | |
| 54 void AudioTrack::AddSink(AudioTrackSinkInterface* sink) { | |
| 55 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 56 if (audio_source_) | |
| 57 audio_source_->AddSink(sink); | |
| 58 } | |
| 59 | |
| 60 void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) { | |
| 61 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 62 if (audio_source_) | |
| 63 audio_source_->RemoveSink(sink); | |
| 64 } | |
| 65 | |
| 66 void AudioTrack::OnChanged() { | |
| 67 RTC_DCHECK(thread_checker_.CalledOnValidThread()); | |
| 68 if (audio_source_->state() == MediaSourceInterface::kEnded) { | |
| 69 set_state(kEnded); | |
| 70 } else { | |
| 71 set_state(kLive); | |
| 72 } | |
| 73 } | |
| 74 | |
| 75 } // namespace webrtc | |
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Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
