816134a8aa
This reverts commit 096ad02c02b4bc6c046282b8793ef84d041dd0d8. Reason for revert: Including a fix for the test issue. Original change's description: > Revert "Fix race between enabled() and set_enabled() in VideoTrack." > > This reverts commit 5ffefe9d2d743c66f8a8bcbc5ad9662a3138840a. > > Reason for revert: Breaks Chromium Android browser tests on fyi bots. > > Original change's description: > > Fix race between enabled() and set_enabled() in VideoTrack. > > > > Along the way I introduced VideoSourceBaseGuarded, which is equivalent > > to VideoSourceBase except that it applies thread checks. I found that > > it's easy to use VideoSourceBase incorrectly and in fact there appear > > to be tests that do this. > > > > I made the source object const in VideoTrack, as it already was in > > AudioTrack, and that allowed for making the GetSource() accessors > > bypass the proxy thread hop and give the caller direct access. > > > > Bug: webrtc:12773, b/188139639, webrtc:12780 > > Change-Id: I022175c4239a1306ef54059c131d81411d5124fe > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219160 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > > Commit-Queue: Tommi <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34096} > > TBR=mbonadei@webrtc.org,tommi@webrtc.org,landrey@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I16323d459c76eb6a87cc602a0048f6ee01c81626 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12773 > Bug: b/188139639 > Bug: webrtc:12780 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219637 > Reviewed-by: Evan Shrubsole <eshr@google.com> > Commit-Queue: Evan Shrubsole <eshr@google.com> > Cr-Commit-Position: refs/heads/master@{#34101} # Not skipping CQ checks because this is a reland. Bug: webrtc:12773 Bug: b/188139639 Bug: webrtc:12780 Change-Id: Ib35fe15a6c43de8f286d60aff02b19df1ab76925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219639 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34104}
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 rtc::make_ref_counted<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_RUN_ON(&thread_checker_);
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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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// Callable from any thread.
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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_RUN_ON(&thread_checker_);
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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_RUN_ON(&thread_checker_);
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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_RUN_ON(&thread_checker_);
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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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