webrtc_m130/pc/audio_track.cc
Tomas Gunnarsson fe328ca88a Add several thread checks to RtpSender classes.
Minor related updates to AudioTrack and VideoTrack's sequence checkers.

There's more that can be done (or arguably needs to), but this is
a start.

Bug: none
Change-Id: I3ccf8eb9bbb6bef62b83248a23a68871b9fcd9e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251843
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36021}
2022-02-17 09:00:54 +00:00

72 lines
1.9 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 rtc::make_ref_counted<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_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