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https://github.com/mollyim/webrtc.git
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Revert "Reland "Remove stopped_
from AudioRtpReceiver and VideoRtpReceiver.""
This reverts commit3ed36c0521
. Reason for revert: Breaks downstream project. Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland ofbb57e2d7aa
> > 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}
This commit is contained in:
parent
b02220d1a0
commit
2da85916ab
10 changed files with 87 additions and 44 deletions
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@ -1416,7 +1416,6 @@ rtc_library("video_track_source") {
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"../media:rtc_media_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:no_unique_address",
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"../rtc_base/system:rtc_export",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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@ -61,6 +61,7 @@ AudioRtpReceiver::AudioRtpReceiver(
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AudioRtpReceiver::~AudioRtpReceiver() {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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RTC_DCHECK(stopped_);
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RTC_DCHECK(!media_channel_);
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track_->GetSource()->UnregisterAudioObserver(this);
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@ -84,10 +85,6 @@ void AudioRtpReceiver::OnChanged() {
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void AudioRtpReceiver::SetOutputVolume_w(double volume) {
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RTC_DCHECK_GE(volume, 0.0);
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RTC_DCHECK_LE(volume, 10.0);
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if (!media_channel_)
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return;
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ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
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: media_channel_->SetDefaultOutputVolume(volume);
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}
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@ -97,10 +94,13 @@ void AudioRtpReceiver::OnSetVolume(double volume) {
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RTC_DCHECK_GE(volume, 0);
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RTC_DCHECK_LE(volume, 10);
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// Update the cached_volume_ even when stopped, to allow clients to set the
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// Update the cached_volume_ even when stopped_, to allow clients to set the
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// volume before starting/restarting, eg see crbug.com/1272566.
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cached_volume_ = volume;
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if (stopped_)
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return;
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// When the track is disabled, the volume of the source, which is the
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// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
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// setting the volume to the source when the track is disabled.
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@ -160,7 +160,10 @@ AudioRtpReceiver::GetFrameDecryptor() const {
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void AudioRtpReceiver::Stop() {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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// TODO(deadbeef): Need to do more here to fully stop receiving packets.
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source_->SetState(MediaSourceInterface::kEnded);
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if (!stopped_) {
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source_->SetState(MediaSourceInterface::kEnded);
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stopped_ = true;
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}
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
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RTC_DCHECK_RUN_ON(worker_thread_);
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@ -180,17 +183,22 @@ void AudioRtpReceiver::StopAndEndTrack() {
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void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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MediaSourceInterface::SourceState state = source_->state();
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worker_thread_->Invoke<void>(
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RTC_FROM_HERE,
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[&, enabled = cached_track_enabled_, volume = cached_volume_]() {
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bool ok = worker_thread_->Invoke<bool>(
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RTC_FROM_HERE, [&, enabled = cached_track_enabled_,
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volume = cached_volume_, was_stopped = stopped_]() {
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RTC_DCHECK_RUN_ON(worker_thread_);
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if (!media_channel_)
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return; // Can't restart.
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if (!media_channel_) {
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RTC_DCHECK(was_stopped);
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return false; // Can't restart.
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}
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if (state != MediaSourceInterface::kInitializing) {
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if (ssrc_ == ssrc)
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return;
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if (!was_stopped && ssrc_ == ssrc) {
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// Already running with that ssrc.
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RTC_DCHECK(worker_thread_safety_->alive());
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return true;
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}
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if (!was_stopped) {
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source_->Stop(media_channel_, ssrc_);
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}
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@ -201,8 +209,13 @@ void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
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}
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Reconfigure(enabled, volume);
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return true;
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});
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source_->SetState(MediaSourceInterface::kLive);
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if (!ok)
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return;
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stopped_ = false;
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}
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void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
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@ -322,6 +335,9 @@ void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
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RTC_DCHECK(media_channel == nullptr ||
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media_channel->media_type() == media_type());
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if (stopped_ && !media_channel)
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return;
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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SetMediaChannel_w(media_channel);
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@ -133,6 +133,7 @@ class AudioRtpReceiver : public ObserverInterface,
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RTC_GUARDED_BY(&signaling_thread_checker_);
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bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);
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double cached_volume_ RTC_GUARDED_BY(&signaling_thread_checker_) = 1.0;
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bool stopped_ RTC_GUARDED_BY(&signaling_thread_checker_) = true;
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RtpReceiverObserverInterface* observer_
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RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
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bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
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@ -63,7 +63,6 @@ TEST_F(AudioRtpReceiverTest, SetOutputVolumeIsCalled) {
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receiver_->track();
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receiver_->track()->set_enabled(true);
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receiver_->SetMediaChannel(&media_channel_);
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EXPECT_CALL(media_channel_, SetDefaultRawAudioSink(_)).Times(0);
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receiver_->SetupMediaChannel(kSsrc);
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EXPECT_CALL(media_channel_, SetOutputVolume(kSsrc, kVolume))
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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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@ -49,11 +49,12 @@ VideoRtpReceiver::VideoRtpReceiver(
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attachment_id_(GenerateUniqueId()) {
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RTC_DCHECK(worker_thread_);
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SetStreams(streams);
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RTC_DCHECK_EQ(source_->state(), MediaSourceInterface::kInitializing);
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RTC_DCHECK_EQ(source_->state(), MediaSourceInterface::kLive);
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}
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VideoRtpReceiver::~VideoRtpReceiver() {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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RTC_DCHECK(stopped_);
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RTC_DCHECK(!media_channel_);
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}
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@ -115,7 +116,10 @@ void VideoRtpReceiver::Stop() {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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// TODO(deadbeef): Need to do more here to fully stop receiving packets.
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source_->SetState(MediaSourceInterface::kEnded);
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if (!stopped_) {
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source_->SetState(MediaSourceInterface::kEnded);
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stopped_ = true;
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}
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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@ -136,30 +140,34 @@ void VideoRtpReceiver::StopAndEndTrack() {
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void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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MediaSourceInterface::SourceState state = source_->state();
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// `stopped_` will be `true` on construction. RestartMediaChannel
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// can in this case function like "ensure started" and flip `stopped_`
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// to false.
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// TODO(tommi): Can we restart the media channel without blocking?
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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bool ok = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&, was_stopped =
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stopped_] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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if (!media_channel_) {
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// Ignore further negotiations if we've already been stopped and don't
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// have an associated media channel.
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return; // Can't restart.
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RTC_DCHECK(was_stopped);
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return false; // Can't restart.
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}
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const bool encoded_sink_enabled = saved_encoded_sink_enabled_;
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if (!was_stopped && ssrc_ == ssrc) {
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// Already running with that ssrc.
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return true;
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}
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if (state != MediaSourceInterface::kInitializing) {
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if (ssrc == ssrc_)
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return;
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// Disconnect from a previous ssrc.
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// Disconnect from the previous ssrc.
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if (!was_stopped) {
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SetSink(nullptr);
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if (encoded_sink_enabled)
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SetEncodedSinkEnabled(false);
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}
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bool encoded_sink_enabled = saved_encoded_sink_enabled_;
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SetEncodedSinkEnabled(false);
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// Set up the new ssrc.
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ssrc_ = std::move(ssrc);
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SetSink(source_->sink());
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@ -179,8 +187,14 @@ void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
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media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
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}
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return true;
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});
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source_->SetState(MediaSourceInterface::kLive);
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if (!ok)
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return;
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stopped_ = false;
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}
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// RTC_RUN_ON(worker_thread_)
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@ -274,6 +288,9 @@ void VideoRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
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RTC_DCHECK(media_channel == nullptr ||
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media_channel->media_type() == media_type());
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if (stopped_ && !media_channel)
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return;
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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SetMediaChannel_w(media_channel);
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@ -141,6 +141,8 @@ class VideoRtpReceiver : public RtpReceiverInternal {
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rtc::Thread* const worker_thread_;
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const std::string id_;
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// See documentation for `stopped_` below for when a valid media channel
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// has been assigned and when this pointer will be null.
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cricket::VideoMediaChannel* media_channel_ RTC_GUARDED_BY(worker_thread_) =
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nullptr;
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absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(worker_thread_);
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@ -150,6 +152,15 @@ class VideoRtpReceiver : public RtpReceiverInternal {
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const rtc::scoped_refptr<VideoTrackProxyWithInternal<VideoTrack>> track_;
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std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_
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RTC_GUARDED_BY(&signaling_thread_checker_);
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// `stopped` is state that's used on the signaling thread to indicate whether
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// a valid `media_channel_` has been assigned and configured. When an instance
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// of VideoRtpReceiver is initially created, `stopped_` is true and will
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// remain true until either `SetupMediaChannel` or
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// `SetupUnsignaledMediaChannel` is called after assigning a media channel.
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// After that, `stopped_` will remain false until `Stop()` is called.
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// Note, for checking the state of the class on the worker thread,
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// check `media_channel_` instead, as that's the main worker thread state.
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bool stopped_ RTC_GUARDED_BY(&signaling_thread_checker_) = true;
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RtpReceiverObserverInterface* observer_
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RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
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bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
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@ -169,6 +169,7 @@ TEST_F(VideoRtpReceiverTest, BroadcastsEncodedFramesWhenEnabled) {
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TEST_F(VideoRtpReceiverTest, EnablesEncodedOutputOnChannelRestart) {
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InSequence s;
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EXPECT_CALL(channel_, ClearRecordableEncodedFrameCallback(0));
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MockVideoSink sink;
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Source()->AddEncodedSink(&sink);
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EXPECT_CALL(channel_, SetRecordableEncodedFrameCallback(4711, _));
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@ -15,12 +15,11 @@
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namespace webrtc {
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VideoTrackSource::VideoTrackSource(bool remote)
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: state_(kInitializing), remote_(remote) {
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: state_(kLive), remote_(remote) {
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worker_thread_checker_.Detach();
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}
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void VideoTrackSource::SetState(SourceState new_state) {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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if (state_ != new_state) {
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state_ = new_state;
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FireOnChanged();
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@ -20,7 +20,6 @@
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#include "api/video/video_sink_interface.h"
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#include "api/video/video_source_interface.h"
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#include "media/base/media_channel.h"
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#include "rtc_base/system/no_unique_address.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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@ -32,10 +31,7 @@ class RTC_EXPORT VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
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explicit VideoTrackSource(bool remote);
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void SetState(SourceState new_state);
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SourceState state() const override {
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RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
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return state_;
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}
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SourceState state() const override { return state_; }
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bool remote() const override { return remote_; }
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bool is_screencast() const override { return false; }
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@ -60,9 +56,8 @@ class RTC_EXPORT VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
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virtual rtc::VideoSourceInterface<VideoFrame>* source() = 0;
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private:
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RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
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RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
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SourceState state_ RTC_GUARDED_BY(&signaling_thread_checker_);
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SequenceChecker worker_thread_checker_;
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SourceState state_;
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const bool remote_;
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};
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