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 1537 matching lines @@
   // 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);
   }
+  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);
+  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);
   }
+  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);
+  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