fc83cdc819
This bypasses the proxy for the following properties: * MediaStream::id() * AudioTrack::kind() and AudioTrack::id() * VideoTrack::kind() and VideoTrack::id() * RtpReceiver::media_type() and RtpReceiver::id() * RtpSender::media_type() and RtpSender::id() * VideoTrackSource::remote() and VideoTrackSource::is_screencast() * RtpTransceiver::media_type() Bug: webrtc:11923 Change-Id: If7edea1781f778af3775515fc4af9a9e151c8103 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183767 Reviewed-by: Chen Xing <chxg@google.com> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32071}
72 lines
1.9 KiB
C++
72 lines
1.9 KiB
C++
/*
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* Copyright 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/audio_track.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/ref_counted_object.h"
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namespace webrtc {
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// static
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rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
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const std::string& id,
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const rtc::scoped_refptr<AudioSourceInterface>& source) {
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return new rtc::RefCountedObject<AudioTrack>(id, source);
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}
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AudioTrack::AudioTrack(const std::string& label,
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const rtc::scoped_refptr<AudioSourceInterface>& source)
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: MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
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if (audio_source_) {
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audio_source_->RegisterObserver(this);
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OnChanged();
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}
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}
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AudioTrack::~AudioTrack() {
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RTC_DCHECK(thread_checker_.IsCurrent());
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set_state(MediaStreamTrackInterface::kEnded);
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if (audio_source_)
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audio_source_->UnregisterObserver(this);
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}
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std::string AudioTrack::kind() const {
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return kAudioKind;
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}
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AudioSourceInterface* AudioTrack::GetSource() const {
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RTC_DCHECK(thread_checker_.IsCurrent());
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return audio_source_.get();
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}
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void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(thread_checker_.IsCurrent());
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if (audio_source_)
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audio_source_->AddSink(sink);
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}
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void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK(thread_checker_.IsCurrent());
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if (audio_source_)
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audio_source_->RemoveSink(sink);
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}
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void AudioTrack::OnChanged() {
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RTC_DCHECK(thread_checker_.IsCurrent());
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if (audio_source_->state() == MediaSourceInterface::kEnded) {
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set_state(kEnded);
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} else {
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set_state(kLive);
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}
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}
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} // namespace webrtc
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