Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.

webrtc/api/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 378 matching lines @@
     case PeerConnectionInterface::kNoHost:
       return (cricket::CF_ALL & ~cricket::CF_HOST);
     case PeerConnectionInterface::kAll:
       return cricket::CF_ALL;
     default:
       ASSERT(false);
   }
   return cricket::CF_NONE;
 }
 
-// Helper method to set a voice/video channel on all applicable senders
-// and receivers when one is created/destroyed by WebRtcSession.
-//
-// Used by On(Voice|Video)Channel(Created|Destroyed)
-template <class SENDER,
-          class RECEIVER,
-          class CHANNEL,
-          class SENDERS,
-          class RECEIVERS>
-void SetChannelOnSendersAndReceivers(CHANNEL* channel,
-                                     SENDERS& senders,
-                                     RECEIVERS& receivers,
-                                     cricket::MediaType media_type) {
-  for (auto& sender : senders) {
-    if (sender->media_type() == media_type) {
-      static_cast<SENDER*>(sender->internal())->SetChannel(channel);
-    }
-  }
-  for (auto& receiver : receivers) {
-    if (receiver->media_type() == media_type) {
-      if (!channel) {
-        receiver->internal()->Stop();
-      }
-      static_cast<RECEIVER*>(receiver->internal())->SetChannel(channel);
-    }
-  }
-}
-
 }  // namespace
 
 namespace webrtc {
 
 // Generate a RTCP CNAME when a PeerConnection is created.
 std::string GenerateRtcpCname() {
   std::string cname;
   if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
     LOG(LS_ERROR) << "Failed to generate CNAME.";
     RTC_DCHECK(false);
@@ skipping 192 matching lines @@
   // Initialize the WebRtcSession. It creates transport channels etc.
   if (!session_->Initialize(factory_->options(), std::move(cert_generator),
                             configuration)) {
     return false;
   }
 
   // Register PeerConnection as receiver of local ice candidates.
   // All the callbacks will be posted to the application from PeerConnection.
   session_->RegisterIceObserver(this);
   session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
-  session_->SignalVoiceChannelCreated.connect(
-      this, &PeerConnection::OnVoiceChannelCreated);
   session_->SignalVoiceChannelDestroyed.connect(
       this, &PeerConnection::OnVoiceChannelDestroyed);
-  session_->SignalVideoChannelCreated.connect(
-      this, &PeerConnection::OnVideoChannelCreated);
   session_->SignalVideoChannelDestroyed.connect(
       this, &PeerConnection::OnVideoChannelDestroyed);
   session_->SignalDataChannelCreated.connect(
       this, &PeerConnection::OnDataChannelCreated);
   session_->SignalDataChannelDestroyed.connect(
       this, &PeerConnection::OnDataChannelDestroyed);
   session_->SignalDataChannelOpenMessage.connect(
       this, &PeerConnection::OnDataChannelOpenMessage);
   return true;
 }
@@ skipping 83 matching lines @@
     LOG(LS_ERROR) << "Sender for track " << track->id() << " already exists.";
     return nullptr;
   }
 
   // TODO(deadbeef): Support adding a track to multiple streams.
   rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
   if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
         signaling_thread(),
         new AudioRtpSender(static_cast<AudioTrackInterface*>(track),
-                           session_->voice_channel(), stats_.get()));
+                           session_.get(), stats_.get()));
     if (!streams.empty()) {
       new_sender->internal()->set_stream_id(streams[0]->label());
     }
     const TrackInfo* track_info = FindTrackInfo(
         local_audio_tracks_, new_sender->internal()->stream_id(), track->id());
     if (track_info) {
       new_sender->internal()->SetSsrc(track_info->ssrc);
     }
   } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
         signaling_thread(),
         new VideoRtpSender(static_cast<VideoTrackInterface*>(track),
-                           session_->video_channel()));
+                           session_.get()));
     if (!streams.empty()) {
       new_sender->internal()->set_stream_id(streams[0]->label());
     }
     const TrackInfo* track_info = FindTrackInfo(
         local_video_tracks_, new_sender->internal()->stream_id(), track->id());
     if (track_info) {
       new_sender->internal()->SetSsrc(track_info->ssrc);
     }
   } else {
     LOG(LS_ERROR) << "CreateSender called with invalid kind: " << track->kind();
@@ skipping 44 matching lines @@
   return DtmfSenderProxy::Create(signaling_thread(), sender.get());
 }
 
 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
     const std::string& kind,
     const std::string& stream_id) {
   TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
   rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
   if (kind == MediaStreamTrackInterface::kAudioKind) {
     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(),
-        new AudioRtpSender(session_->voice_channel(), stats_.get()));
+        signaling_thread(), new AudioRtpSender(session_.get(), stats_.get()));
   } else if (kind == MediaStreamTrackInterface::kVideoKind) {
     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(), new VideoRtpSender(session_->video_channel()));
+        signaling_thread(), new VideoRtpSender(session_.get()));
   } else {
     LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
     return new_sender;
   }
   if (!stream_id.empty()) {
     new_sender->internal()->set_stream_id(stream_id);
   }
   senders_.push_back(new_sender);
   return new_sender;
 }
@@ skipping 507 matching lines @@
       RTC_DCHECK(false && "Not implemented");
       break;
   }
 }
 
 void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
                                          const std::string& track_id,
                                          uint32_t ssrc) {
   receivers_.push_back(
       RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
-          signaling_thread(), new AudioRtpReceiver(stream, track_id, ssrc,
-                                                   session_->voice_channel())));
+          signaling_thread(),
+          new AudioRtpReceiver(stream, track_id, ssrc, session_.get())));
 }
 
 void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
                                          const std::string& track_id,
                                          uint32_t ssrc) {
   receivers_.push_back(
       RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
           signaling_thread(),
           new VideoRtpReceiver(stream, track_id, factory_->worker_thread(),
-                               ssrc, session_->video_channel())));
+                               ssrc, session_.get())));
 }
 
 // TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
 // description.
 void PeerConnection::DestroyReceiver(const std::string& track_id) {
   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::StopReceivers(cricket::MediaType media_type) {
+  TrackInfos* current_tracks = GetRemoteTracks(media_type);
+  for (const auto& track_info : *current_tracks) {
+    auto it = FindReceiverForTrack(track_info.track_id);
+    if (it == receivers_.end()) {
+      LOG(LS_WARNING) << "RtpReceiver for track with id " << track_info.track_id
+                      << " doesn't exist.";
+    } else {
+      (*it)->internal()->Stop();
+    }
+  }
+}
+
 void PeerConnection::OnIceConnectionChange(
     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 45 matching lines @@
   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()));
+          signaling_thread(), new AudioRtpSender(track, stream->label(),
+                                                 session_.get(), 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) {
@@ skipping 21 matching lines @@
   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()));
+          signaling_thread(),
+          new VideoRtpSender(track, stream->label(), session_.get()));
   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::OnVideoTrackRemoved(VideoTrackInterface* track,
                                          MediaStreamInterface* stream) {
@@ skipping 521 matching lines @@
       // we can't free it directly here; we need to free it asynchronously.
       sctp_data_channels_to_free_.push_back(*it);
       sctp_data_channels_.erase(it);
       signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
                                nullptr);
       return;
     }
   }
 }
 
-void PeerConnection::OnVoiceChannelCreated() {
-  SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver>(
-      session_->voice_channel(), senders_, receivers_,
-      cricket::MEDIA_TYPE_AUDIO);
-}
-
 void PeerConnection::OnVoiceChannelDestroyed() {
-  SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver,
-                                  cricket::VoiceChannel>(
-      nullptr, senders_, receivers_, cricket::MEDIA_TYPE_AUDIO);
-}
-
-void PeerConnection::OnVideoChannelCreated() {
-  SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver>(
-      session_->video_channel(), senders_, receivers_,
-      cricket::MEDIA_TYPE_VIDEO);
+  StopReceivers(cricket::MEDIA_TYPE_AUDIO);
 }
 
 void PeerConnection::OnVideoChannelDestroyed() {
-  SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver,
-                                  cricket::VideoChannel>(
-      nullptr, senders_, receivers_, cricket::MEDIA_TYPE_VIDEO);
+  StopReceivers(cricket::MEDIA_TYPE_VIDEO);
 }
 
 void PeerConnection::OnDataChannelCreated() {
   for (const auto& channel : sctp_data_channels_) {
     channel->OnTransportChannelCreated();
   }
 }
 
 void PeerConnection::OnDataChannelDestroyed() {
   // Use a temporary copy of the RTP/SCTP DataChannel list because the
@@ skipping 155 matching lines @@
   port_allocator_->set_candidate_filter(
       ConvertIceTransportTypeToCandidateFilter(configuration.type));
   // Call this last since it may create pooled allocator sessions using the
   // candidate filter set above.
   port_allocator_->SetConfiguration(stun_servers, turn_servers,
                                     configuration.ice_candidate_pool_size);
   return true;
 }
 
 }  // namespace webrtc