7bb87ee4e8
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection to //webrtc/pc:libjingle_peerconnection. Moved the RTCStatsCollectorCallback into its own header file, so that PeerConnectionInterface can include that instead of pulling in RTCStatsCollector and PeerConnection and everything. Separated cricket::MediaType into its own header/source set, so that it can be used in the api. BUG=webrtc:5883 Review-Url: https://codereview.webrtc.org/2514883002 Cr-Commit-Position: refs/heads/master@{#16210}
74 lines
2.0 KiB
C++
74 lines
2.0 KiB
C++
/*
|
|
* 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 "webrtc/pc/audiotrack.h"
|
|
|
|
#include "webrtc/base/checks.h"
|
|
|
|
using rtc::scoped_refptr;
|
|
|
|
namespace webrtc {
|
|
|
|
// static
|
|
scoped_refptr<AudioTrack> AudioTrack::Create(
|
|
const std::string& id,
|
|
const scoped_refptr<AudioSourceInterface>& source) {
|
|
return new rtc::RefCountedObject<AudioTrack>(id, source);
|
|
}
|
|
|
|
AudioTrack::AudioTrack(const std::string& label,
|
|
const 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
|