webrtc_m130/webrtc/pc/audiotrack.cc
Edward Lemur c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00

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/rtc_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