[Unified Plan] Don't end audio tracks when SSRC changes.
The RemoteAudioSource has an AudioDataProxy that acts as a sink, passing
along data from AudioRecvStreams to the RemoteAudioSource. If an SSRC is
changed (or other reconfiguration happens) with SDP, the recv stream and
proxy get recreated.
In Plan B, because remote tracks maps 1:1 with SSRCs, it made sense to
end remote track/audio source in response to this. In Plan B, a new
receiver, with a new track and a new proxy would be created for the new
SSRC.
In Unified Plan however, remote tracks correspond to m= sections. The
remote track should only end on port:0 (or RTCP BYE or timeout, etc),
not because the recv stream of an m= section is recreated. The code
already supports changing SSRC and this is working correctly, but
because ~AudioDataProxy() would end the source this would cause the
MediaStreamTrack of the receiver to end (even though the media engine
is still processing the remote audio stream correctly under the hood).
This issue only happened on audio tracks, and because of timing of
PostTasks the track would kEnd in Chromium *after* promise.then().
This CL fixes that issue by not ending the source when the proxy is
destroyed. Destroying a recv stream is a temporary action in Unified
Plan, unless stopped. Tests are added ensuring tracks are kLive.
I have manually verified that this CL fixes the issue and that both
audio and video is flowing through the entire pipeline:
https://jsfiddle.net/henbos/h21xec97/122/
Bug: chromium:1121454
Change-Id: Ic21ac8ea263ccf021b96a14d3e4e3b24eb756c86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214136
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33645}
diff --git a/pc/audio_rtp_receiver.cc b/pc/audio_rtp_receiver.cc
index e8fad28..48553ba 100644
--- a/pc/audio_rtp_receiver.cc
+++ b/pc/audio_rtp_receiver.cc
@@ -28,18 +28,25 @@
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
- std::vector<std::string> stream_ids)
+ std::vector<std::string> stream_ids,
+ bool is_unified_plan)
: AudioRtpReceiver(worker_thread,
receiver_id,
- CreateStreamsFromIds(std::move(stream_ids))) {}
+ CreateStreamsFromIds(std::move(stream_ids)),
+ is_unified_plan) {}
AudioRtpReceiver::AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
- const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
+ bool is_unified_plan)
: worker_thread_(worker_thread),
id_(receiver_id),
- source_(new rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
+ source_(new rtc::RefCountedObject<RemoteAudioSource>(
+ worker_thread,
+ is_unified_plan
+ ? RemoteAudioSource::OnAudioChannelGoneAction::kSurvive
+ : RemoteAudioSource::OnAudioChannelGoneAction::kEnd)),
track_(AudioTrackProxyWithInternal<AudioTrack>::Create(
rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
@@ -137,6 +144,7 @@
if (stopped_) {
return;
}
+ source_->SetState(MediaSourceInterface::kEnded);
if (media_channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
diff --git a/pc/audio_rtp_receiver.h b/pc/audio_rtp_receiver.h
index ec77bbc..789d4a0 100644
--- a/pc/audio_rtp_receiver.h
+++ b/pc/audio_rtp_receiver.h
@@ -43,12 +43,14 @@
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
- std::vector<std::string> stream_ids);
+ std::vector<std::string> stream_ids,
+ bool is_unified_plan);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
- const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
+ bool is_unified_plan);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
diff --git a/pc/peer_connection_rtp_unittest.cc b/pc/peer_connection_rtp_unittest.cc
index 3614f0c..54287e7 100644
--- a/pc/peer_connection_rtp_unittest.cc
+++ b/pc/peer_connection_rtp_unittest.cc
@@ -779,6 +779,56 @@
EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId1);
EXPECT_EQ(receivers[0]->streams()[1]->id(), kStreamId2);
}
+TEST_F(PeerConnectionRtpTestUnifiedPlan, TracksDoNotEndWhenSsrcChanges) {
+ constexpr uint32_t kFirstMungedSsrc = 1337u;
+
+ auto caller = CreatePeerConnection();
+ auto callee = CreatePeerConnection();
+
+ // Caller offers to receive audio and video.
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kRecvOnly;
+ caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+
+ // Callee wants to send audio and video tracks.
+ callee->AddTrack(callee->CreateAudioTrack("audio_track"), {});
+ callee->AddTrack(callee->CreateVideoTrack("video_track"), {});
+
+ // Do inittial offer/answer exchange.
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ ASSERT_TRUE(
+ caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
+ ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
+ ASSERT_EQ(caller->pc()->GetReceivers().size(), 2u);
+
+ // Do a follow-up offer/answer exchange where the SSRCs are modified.
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ auto answer = callee->CreateAnswer();
+ auto& contents = answer->description()->contents();
+ ASSERT_TRUE(!contents.empty());
+ for (size_t i = 0; i < contents.size(); ++i) {
+ auto& mutable_streams = contents[i].media_description()->mutable_streams();
+ ASSERT_EQ(mutable_streams.size(), 1u);
+ mutable_streams[0].ssrcs = {kFirstMungedSsrc + static_cast<uint32_t>(i)};
+ }
+ ASSERT_TRUE(
+ callee->SetLocalDescription(CloneSessionDescription(answer.get())));
+ ASSERT_TRUE(
+ caller->SetRemoteDescription(CloneSessionDescription(answer.get())));
+
+ // No furher track events should fire because we never changed direction, only
+ // SSRCs.
+ ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
+ // We should have the same number of receivers as before.
+ auto receivers = caller->pc()->GetReceivers();
+ ASSERT_EQ(receivers.size(), 2u);
+ // The tracks are still alive.
+ EXPECT_EQ(receivers[0]->track()->state(),
+ MediaStreamTrackInterface::TrackState::kLive);
+ EXPECT_EQ(receivers[1]->track()->state(),
+ MediaStreamTrackInterface::TrackState::kLive);
+}
// Tests that with Unified Plan if the the stream id changes for a track when
// when setting a new remote description, that the media stream is updated
diff --git a/pc/remote_audio_source.cc b/pc/remote_audio_source.cc
index 3f53bb8..848fba3 100644
--- a/pc/remote_audio_source.cc
+++ b/pc/remote_audio_source.cc
@@ -49,9 +49,12 @@
const rtc::scoped_refptr<RemoteAudioSource> source_;
};
-RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
+RemoteAudioSource::RemoteAudioSource(
+ rtc::Thread* worker_thread,
+ OnAudioChannelGoneAction on_audio_channel_gone_action)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
+ on_audio_channel_gone_action_(on_audio_channel_gone_action),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
@@ -90,6 +93,13 @@
});
}
+void RemoteAudioSource::SetState(SourceState new_state) {
+ if (state_ != new_state) {
+ state_ = new_state;
+ FireOnChanged();
+ }
+}
+
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
@@ -156,6 +166,9 @@
}
void RemoteAudioSource::OnAudioChannelGone() {
+ if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
+ return;
+ }
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
// This object needs to live long enough for the cleanup logic in OnMessage to
@@ -170,8 +183,7 @@
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
- state_ = MediaSourceInterface::kEnded;
- FireOnChanged();
+ SetState(MediaSourceInterface::kEnded);
// Will possibly delete this RemoteAudioSource since it is reference counted
// in the message.
delete msg->pdata;
diff --git a/pc/remote_audio_source.h b/pc/remote_audio_source.h
index 276a103..2eae073 100644
--- a/pc/remote_audio_source.h
+++ b/pc/remote_audio_source.h
@@ -40,7 +40,21 @@
class RemoteAudioSource : public Notifier<AudioSourceInterface>,
rtc::MessageHandler {
public:
- explicit RemoteAudioSource(rtc::Thread* worker_thread);
+ // In Unified Plan, receivers map to m= sections and their tracks and sources
+ // survive SSRCs being reconfigured. The life cycle of the remote audio source
+ // is associated with the life cycle of the m= section, and thus even if an
+ // audio channel is destroyed the RemoteAudioSource should kSurvive.
+ //
+ // In Plan B however, remote audio sources map 1:1 with an SSRCs and if an
+ // audio channel is destroyed, the RemoteAudioSource should kEnd.
+ enum class OnAudioChannelGoneAction {
+ kSurvive,
+ kEnd,
+ };
+
+ explicit RemoteAudioSource(
+ rtc::Thread* worker_thread,
+ OnAudioChannelGoneAction on_audio_channel_gone_action);
// Register and unregister remote audio source with the underlying media
// engine.
@@ -48,6 +62,7 @@
absl::optional<uint32_t> ssrc);
void Stop(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc);
+ void SetState(SourceState new_state);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
@@ -75,6 +90,7 @@
rtc::Thread* const main_thread_;
rtc::Thread* const worker_thread_;
+ const OnAudioChannelGoneAction on_audio_channel_gone_action_;
std::list<AudioObserver*> audio_observers_;
Mutex sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index 4d6d58d..97093e8 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -302,7 +302,8 @@
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ =
- new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams);
+ new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams,
+ /*is_unified_plan=*/true);
audio_rtp_receiver_->SetMediaChannel(voice_media_channel_);
audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
audio_track_ = audio_rtp_receiver_->audio_track();
diff --git a/pc/rtp_transmission_manager.cc b/pc/rtp_transmission_manager.cc
index e796f9b..eaf29b8 100644
--- a/pc/rtp_transmission_manager.cc
+++ b/pc/rtp_transmission_manager.cc
@@ -240,8 +240,9 @@
receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
- signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
- std::vector<std::string>({})));
+ signaling_thread(),
+ new AudioRtpReceiver(worker_thread(), receiver_id,
+ std::vector<std::string>({}), IsUnifiedPlan()));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
@@ -453,7 +454,7 @@
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* audio_receiver = new AudioRtpReceiver(
- worker_thread(), remote_sender_info.sender_id, streams);
+ worker_thread(), remote_sender_info.sender_id, streams, IsUnifiedPlan());
audio_receiver->SetMediaChannel(voice_media_channel());
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
audio_receiver->SetupUnsignaledMediaChannel();
