| /* |
| * Copyright 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/audio_track.h" |
| |
| #include "rtc_base/checks.h" |
| #include "rtc_base/ref_counted_object.h" |
| |
| namespace webrtc { |
| |
| // static |
| rtc::scoped_refptr<AudioTrack> AudioTrack::Create( |
| const std::string& id, |
| const rtc::scoped_refptr<AudioSourceInterface>& source) { |
| return new rtc::RefCountedObject<AudioTrack>(id, source); |
| } |
| |
| AudioTrack::AudioTrack(const std::string& label, |
| const rtc::scoped_refptr<AudioSourceInterface>& source) |
| : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) { |
| if (audio_source_) { |
| audio_source_->RegisterObserver(this); |
| OnChanged(); |
| } |
| } |
| |
| AudioTrack::~AudioTrack() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| set_state(MediaStreamTrackInterface::kEnded); |
| if (audio_source_) |
| audio_source_->UnregisterObserver(this); |
| } |
| |
| std::string AudioTrack::kind() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return kAudioKind; |
| } |
| |
| AudioSourceInterface* AudioTrack::GetSource() const { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| return audio_source_.get(); |
| } |
| |
| void AudioTrack::AddSink(AudioTrackSinkInterface* sink) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| if (audio_source_) |
| audio_source_->AddSink(sink); |
| } |
| |
| void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| if (audio_source_) |
| audio_source_->RemoveSink(sink); |
| } |
| |
| void AudioTrack::OnChanged() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| if (audio_source_->state() == MediaSourceInterface::kEnded) { |
| set_state(kEnded); |
| } else { |
| set_state(kLive); |
| } |
| } |
| |
| } // namespace webrtc |