When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded

webrtc/pc/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 528 matching lines @@
                                           &PeerConnection::OnAudioTrackAdded);
   observer->SignalAudioTrackRemoved.connect(
       this, &PeerConnection::OnAudioTrackRemoved);
   observer->SignalVideoTrackAdded.connect(this,
                                           &PeerConnection::OnVideoTrackAdded);
   observer->SignalVideoTrackRemoved.connect(
       this, &PeerConnection::OnVideoTrackRemoved);
   stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
 
   for (const auto& track : local_stream->GetAudioTracks()) {
-    OnAudioTrackAdded(track.get(), local_stream);
+    AddAudioTrack(track.get(), local_stream);
   }
   for (const auto& track : local_stream->GetVideoTracks()) {
-    OnVideoTrackAdded(track.get(), local_stream);
+    AddVideoTrack(track.get(), local_stream);
   }
 
   stats_->AddStream(local_stream);
   observer_->OnRenegotiationNeeded();
   return true;
 }
 
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
   TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
-  for (const auto& track : local_stream->GetAudioTracks()) {
-    OnAudioTrackRemoved(track.get(), local_stream);
+  if (!IsClosed()) {
+    for (const auto& track : local_stream->GetAudioTracks()) {
+      RemoveAudioTrack(track.get(), local_stream);
+    }
+    for (const auto& track : local_stream->GetVideoTracks()) {
+      RemoveVideoTrack(track.get(), local_stream);
+    }
   }
-  for (const auto& track : local_stream->GetVideoTracks()) {
-    OnVideoTrackRemoved(track.get(), local_stream);
-  }
-
   local_streams_->RemoveStream(local_stream);
   stream_observers_.erase(
       std::remove_if(
           stream_observers_.begin(), stream_observers_.end(),
           [local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
             return observer->stream()->label().compare(local_stream->label()) ==
                    0;
           }),
       stream_observers_.end());
 
@@ skipping 876 matching lines @@
   auto it = FindReceiverForTrack(track_id);
   if (it == receivers_.end()) {
     LOG(LS_WARNING) << "RtpReceiver for track with id " << track_id
                     << " doesn't exist.";
   } else {
     (*it)->internal()->Stop();
     receivers_.erase(it);
   }
 }
 
+void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
+                                   MediaStreamInterface* stream) {
+  RTC_DCHECK(!IsClosed());
+  auto sender = FindSenderForTrack(track);
+  if (sender != senders_.end()) {
+    // We already have a sender for this track, so just change the stream_id
+    // so that it's correct in the next call to CreateOffer.
+    (*sender)->internal()->set_stream_id(stream->label());
+    return;
+  }
+
+  // Normal case; we've never seen this track before.
+  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
+      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
+          signaling_thread(),
+          new AudioRtpSender(track, stream->label(), session_->voice_channel(),
+                             stats_.get()));
+  senders_.push_back(new_sender);
+  // If the sender has already been configured in SDP, we call SetSsrc,
+  // which will connect the sender to the underlying transport. This can
+  // occur if a local session description that contains the ID of the sender
+  // is set before AddStream is called. It can also occur if the local
+  // session description is not changed and RemoveStream is called, and
+  // later AddStream is called again with the same stream.
+  const TrackInfo* track_info =
+      FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
+  if (track_info) {
+    new_sender->internal()->SetSsrc(track_info->ssrc);
+  }
+}
+
+// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
+// indefinitely, when we have unified plan SDP.
+void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
+                                      MediaStreamInterface* stream) {
+  RTC_DCHECK(!IsClosed());
+  auto sender = FindSenderForTrack(track);
+  if (sender == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+                    << " doesn't exist.";
+    return;
+  }
+  (*sender)->internal()->Stop();
+  senders_.erase(sender);
+}
+
+void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
+                                   MediaStreamInterface* stream) {
+  RTC_DCHECK(!IsClosed());
+  auto sender = FindSenderForTrack(track);
+  if (sender != senders_.end()) {
+    // We already have a sender for this track, so just change the stream_id
+    // so that it's correct in the next call to CreateOffer.
+    (*sender)->internal()->set_stream_id(stream->label());
+    return;
+  }
+
+  // Normal case; we've never seen this track before.
+  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
+      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
+          signaling_thread(), new VideoRtpSender(track, stream->label(),
+                                                 session_->video_channel()));
+  senders_.push_back(new_sender);
+  const TrackInfo* track_info =
+      FindTrackInfo(local_video_tracks_, stream->label(), track->id());
+  if (track_info) {
+    new_sender->internal()->SetSsrc(track_info->ssrc);
+  }
+}
+
+void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
+                                      MediaStreamInterface* stream) {
+  RTC_DCHECK(!IsClosed());
+  auto sender = FindSenderForTrack(track);
+  if (sender == senders_.end()) {
+    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+                    << " doesn't exist.";
+    return;
+  }
+  (*sender)->internal()->Stop();
+  senders_.erase(sender);
+}
+
 void PeerConnection::OnIceConnectionStateChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // After transitioning to "closed", ignore any additional states from
   // WebRtcSession (such as "disconnected").
   if (IsClosed()) {
     return;
   }
   ice_connection_state_ = new_state;
   observer_->OnIceConnectionChange(ice_connection_state_);
@@ skipping 47 matching lines @@
     }
   }
   observer_->OnSignalingChange(signaling_state_);
 }
 
 void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
                                        MediaStreamInterface* stream) {
   if (IsClosed()) {
     return;
   }
-  auto sender = FindSenderForTrack(track);
-  if (sender != senders_.end()) {
-    // We already have a sender for this track, so just change the stream_id
-    // so that it's correct in the next call to CreateOffer.
-    (*sender)->internal()->set_stream_id(stream->label());
-    return;
-  }
-
-  // Normal case; we've never seen this track before.
-  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
-      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-          signaling_thread(),
-          new AudioRtpSender(track, stream->label(), session_->voice_channel(),
-                             stats_.get()));
-  senders_.push_back(new_sender);
-  // If the sender has already been configured in SDP, we call SetSsrc,
-  // which will connect the sender to the underlying transport. This can
-  // occur if a local session description that contains the ID of the sender
-  // is set before AddStream is called. It can also occur if the local
-  // session description is not changed and RemoveStream is called, and
-  // later AddStream is called again with the same stream.
-  const TrackInfo* track_info =
-      FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
-  if (track_info) {
-    new_sender->internal()->SetSsrc(track_info->ssrc);
-  }
+  AddAudioTrack(track, stream);
+  observer_->OnRenegotiationNeeded();
 }
 
-// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
 void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
                                          MediaStreamInterface* stream) {
   if (IsClosed()) {
     return;
   }
-  auto sender = FindSenderForTrack(track);
-  if (sender == senders_.end()) {
-    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                    << " doesn't exist.";
-    return;
-  }
-  (*sender)->internal()->Stop();
-  senders_.erase(sender);
+  RemoveAudioTrack(track, stream);
+  observer_->OnRenegotiationNeeded();
 }
 
 void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
                                        MediaStreamInterface* stream) {
   if (IsClosed()) {
     return;
   }
-  auto sender = FindSenderForTrack(track);
-  if (sender != senders_.end()) {
-    // We already have a sender for this track, so just change the stream_id
-    // so that it's correct in the next call to CreateOffer.
-    (*sender)->internal()->set_stream_id(stream->label());
-    return;
-  }
-
-  // Normal case; we've never seen this track before.
-  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
-      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-          signaling_thread(), new VideoRtpSender(track, stream->label(),
-                                                 session_->video_channel()));
-  senders_.push_back(new_sender);
-  const TrackInfo* track_info =
-      FindTrackInfo(local_video_tracks_, stream->label(), track->id());
-  if (track_info) {
-    new_sender->internal()->SetSsrc(track_info->ssrc);
-  }
+  AddVideoTrack(track, stream);
+  observer_->OnRenegotiationNeeded();
 }
 
 void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
                                          MediaStreamInterface* stream) {
   if (IsClosed()) {
     return;
   }
-  auto sender = FindSenderForTrack(track);
-  if (sender == senders_.end()) {
-    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                    << " doesn't exist.";
-    return;
-  }
-  (*sender)->internal()->Stop();
-  senders_.erase(sender);
+  RemoveVideoTrack(track, stream);
+  observer_->OnRenegotiationNeeded();
 }
 
 void PeerConnection::PostSetSessionDescriptionFailure(
     SetSessionDescriptionObserver* observer,
     const std::string& error) {
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   msg->error = error;
   signaling_thread()->Post(RTC_FROM_HERE, this,
                            MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
 }
@@ skipping 742 matching lines @@
   return event_log_->StartLogging(file, max_size_bytes);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   if (event_log_) {
     event_log_->StopLogging();
   }
 }
 
 }  // namespace webrtc