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/*
* 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"
namespace webrtc {
// static
rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
absl::string_view id,
const rtc::scoped_refptr<AudioSourceInterface>& source) {
return rtc::make_ref_counted<AudioTrack>(id, source);
}
AudioTrack::AudioTrack(absl::string_view 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_RUN_ON(&signaling_thread_checker_);
set_state(MediaStreamTrackInterface::kEnded);
if (audio_source_)
audio_source_->UnregisterObserver(this);
}
std::string AudioTrack::kind() const {
return kAudioKind;
}
AudioSourceInterface* AudioTrack::GetSource() const {
// Callable from any thread.
return audio_source_.get();
}
void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (audio_source_)
audio_source_->AddSink(sink);
}
void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (audio_source_)
audio_source_->RemoveSink(sink);
}
void AudioTrack::OnChanged() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (audio_source_->state() == MediaSourceInterface::kEnded) {
set_state(kEnded);
} else {
set_state(kLive);
}
}
} // namespace webrtc
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