Revert "Reland "Remove stopped_ from AudioRtpReceiver and VideoRtpReceiver.""
This reverts commit 3ed36c0521546881656c73984456485dcab16205. Reason for revert: Breaks downstream project. Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f > > The difference from the original CL is that a check for > `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed. > This caused a side effect that registering the sink while the source > was in an "initializing" state, failed. The last remaining state > however, is `kEnded` - but since there's no logic in the class around > the expected value of the states, the check inside of AddSink() > doesn't provide an additional value - it's rather a surprise for > developers if it doesn't succeed. So, now removed. > > Original change's description: > > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver. > > > > This simplifies the logic in these classes a bit, which makes upcoming > > change easier. The `stopped_` flag in these classes was essentially > > the same thing as `media_channel_ == nullptr`, which is what's > > consistently used now for the same checks. > > > > Bug: webrtc:13540 > > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35907} > > Bug: webrtc:13540 > Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35958} TBR=ilnik@webrtc.org,tommi@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: Ieb7235d88c808c78ad0847403be991d4dce1ace6 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:13540 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251383 Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35963}
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WebRTC LUCI CQ
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@ -55,7 +55,7 @@ RemoteAudioSource::RemoteAudioSource(
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: main_thread_(rtc::Thread::Current()),
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worker_thread_(worker_thread),
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on_audio_channel_gone_action_(on_audio_channel_gone_action),
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state_(MediaSourceInterface::kInitializing) {
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state_(MediaSourceInterface::kLive) {
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RTC_DCHECK(main_thread_);
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RTC_DCHECK(worker_thread_);
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}
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@ -134,6 +134,11 @@ void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
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RTC_DCHECK_RUN_ON(main_thread_);
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RTC_DCHECK(sink);
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if (state_ != MediaSourceInterface::kLive) {
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RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
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return;
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}
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MutexLock lock(&sink_lock_);
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RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
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sinks_.push_back(sink);
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