mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-17 07:37:51 +01:00

All changes outside thread_checker.h are by: s/CalledOnValidThread/IsCurrent/ s/DetachFromThread/Detach/ Bug: webrtc:9883 Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27494}
72 lines
2 KiB
C++
72 lines
2 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 "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_.IsCurrent());
|
|
set_state(MediaStreamTrackInterface::kEnded);
|
|
if (audio_source_)
|
|
audio_source_->UnregisterObserver(this);
|
|
}
|
|
|
|
std::string AudioTrack::kind() const {
|
|
RTC_DCHECK(thread_checker_.IsCurrent());
|
|
return kAudioKind;
|
|
}
|
|
|
|
AudioSourceInterface* AudioTrack::GetSource() const {
|
|
RTC_DCHECK(thread_checker_.IsCurrent());
|
|
return audio_source_.get();
|
|
}
|
|
|
|
void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
|
|
RTC_DCHECK(thread_checker_.IsCurrent());
|
|
if (audio_source_)
|
|
audio_source_->AddSink(sink);
|
|
}
|
|
|
|
void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
|
|
RTC_DCHECK(thread_checker_.IsCurrent());
|
|
if (audio_source_)
|
|
audio_source_->RemoveSink(sink);
|
|
}
|
|
|
|
void AudioTrack::OnChanged() {
|
|
RTC_DCHECK(thread_checker_.IsCurrent());
|
|
if (audio_source_->state() == MediaSourceInterface::kEnded) {
|
|
set_state(kEnded);
|
|
} else {
|
|
set_state(kLive);
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|