Refactoring of RTCP options in BaseChannel.

webrtc/api/webrtcsession.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 1769 matching lines @@
     }
   }
 
   return true;
 }
 
 bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
                                        const std::string* bundle_transport) {
   bool require_rtcp_mux =
       rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
-  bool create_rtcp_transport_channel = !require_rtcp_mux;
 
   std::string transport_name =
       bundle_transport ? *bundle_transport : content->name;
 
   cricket::TransportChannel* rtp_transport =
       transport_controller_->CreateTransportChannel(
           transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   cricket::TransportChannel* rtcp_transport = nullptr;
-  if (create_rtcp_transport_channel) {
+  if (!require_rtcp_mux) {
     rtcp_transport = transport_controller_->CreateTransportChannel(
         transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 
   voice_channel_.reset(channel_manager_->CreateVoiceChannel(
       media_controller_, rtp_transport, rtcp_transport,
       transport_controller_->signaling_thread(), content->name,
-      bundle_transport, create_rtcp_transport_channel, SrtpRequired(),
-      audio_options_));
+      bundle_transport, require_rtcp_mux, SrtpRequired(), audio_options_));
   if (!voice_channel_) {
     return false;
   }
 
-  voice_channel_->SignalDestroyRtcpTransport.connect(
-      this, &WebRtcSession::OnDestroyRtcpTransport_n);
-  if (require_rtcp_mux) {
-    voice_channel_->ActivateRtcpMux();
-  }
-
+  voice_channel_->SignalRtcpMuxFullyActive.connect(
+      this, &WebRtcSession::DestroyRtcpTransport_n);
   voice_channel_->SignalDtlsSrtpSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSrtpSetupFailure);
 
   SignalVoiceChannelCreated();
   voice_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
                                        const std::string* bundle_transport) {
   bool require_rtcp_mux =
       rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
-  bool create_rtcp_transport_channel = !require_rtcp_mux;
 
   std::string transport_name =
       bundle_transport ? *bundle_transport : content->name;
 
   cricket::TransportChannel* rtp_transport =
       transport_controller_->CreateTransportChannel(
           transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   cricket::TransportChannel* rtcp_transport = nullptr;
-  if (create_rtcp_transport_channel) {
+  if (!require_rtcp_mux) {
     rtcp_transport = transport_controller_->CreateTransportChannel(
         transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 
   video_channel_.reset(channel_manager_->CreateVideoChannel(
       media_controller_, rtp_transport, rtcp_transport,
       transport_controller_->signaling_thread(), content->name,
-      bundle_transport, create_rtcp_transport_channel, SrtpRequired(),
-      video_options_));
+      bundle_transport, require_rtcp_mux, SrtpRequired(), video_options_));
 
   if (!video_channel_) {
     return false;
   }
 
-  video_channel_->SignalDestroyRtcpTransport.connect(
-      this, &WebRtcSession::OnDestroyRtcpTransport_n);
-  if (require_rtcp_mux) {
-    video_channel_->ActivateRtcpMux();
-  }
+  video_channel_->SignalRtcpMuxFullyActive.connect(
+      this, &WebRtcSession::DestroyRtcpTransport_n);
   video_channel_->SignalDtlsSrtpSetupFailure.connect(
       this, &WebRtcSession::OnDtlsSrtpSetupFailure);
 
   SignalVideoChannelCreated();
   video_channel_->SignalSentPacket.connect(this,
                                            &WebRtcSession::OnSentPacket_w);
   return true;
 }
 
 bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
@@ skipping 16 matching lines @@
       return false;
     }
     if (!network_thread_->Invoke<bool>(
             RTC_FROM_HERE, rtc::Bind(&WebRtcSession::CreateSctpTransport_n,
                                      this, content->name, transport_name))) {
       return false;
     };
   } else {
     bool require_rtcp_mux =
         rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
-    bool create_rtcp_transport_channel = !sctp && !require_rtcp_mux;
 
     std::string transport_name =
         bundle_transport ? *bundle_transport : content->name;
     cricket::TransportChannel* rtp_transport =
         transport_controller_->CreateTransportChannel(
             transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
     cricket::TransportChannel* rtcp_transport = nullptr;
-    if (create_rtcp_transport_channel) {
+    if (!require_rtcp_mux) {
       rtcp_transport = transport_controller_->CreateTransportChannel(
           transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
     }
 
     rtp_data_channel_.reset(channel_manager_->CreateRtpDataChannel(
         media_controller_, rtp_transport, rtcp_transport,
         transport_controller_->signaling_thread(), content->name,
-        bundle_transport, create_rtcp_transport_channel, SrtpRequired()));
+        bundle_transport, require_rtcp_mux, SrtpRequired()));
 
     if (!rtp_data_channel_) {
       return false;
     }
 
-    rtp_data_channel_->SignalDestroyRtcpTransport.connect(
-        this, &WebRtcSession::OnDestroyRtcpTransport_n);
-
-    if (require_rtcp_mux) {
-      rtp_data_channel_->ActivateRtcpMux();
-    }
+    rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
+        this, &WebRtcSession::DestroyRtcpTransport_n);
     rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
         this, &WebRtcSession::OnDtlsSrtpSetupFailure);
     rtp_data_channel_->SignalSentPacket.connect(this,
                                                 &WebRtcSession::OnSentPacket_w);
   }
 
   SignalDataChannelCreated();
 
   return true;
 }
@@ skipping 447 matching lines @@
 }
 
 void WebRtcSession::DestroyTransport(const std::string& transport_name,
                                      int component) {
   network_thread_->Invoke<void>(
       RTC_FROM_HERE,
       rtc::Bind(&cricket::TransportController::DestroyTransportChannel_n,
                 transport_controller_.get(), transport_name, component));
 }
 
-void WebRtcSession::OnDestroyRtcpTransport_n(
-    const std::string& transport_name) {
+void WebRtcSession::DestroyRtcpTransport_n(const std::string& transport_name) {
   ASSERT(network_thread()->IsCurrent());
   transport_controller_->DestroyTransportChannel_n(
       transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
 }
 
 void WebRtcSession::DestroyVideoChannel() {
   SignalVideoChannelDestroyed();
   RTC_DCHECK(video_channel_->rtp_transport());
   std::string transport_name;
   transport_name = video_channel_->rtp_transport()->transport_name();
@@ skipping 24 matching lines @@
   std::string transport_name;
   transport_name = rtp_data_channel_->rtp_transport()->transport_name();
   bool need_to_delete_rtcp = (rtp_data_channel_->rtcp_transport() != nullptr);
   channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
   DestroyTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
   if (need_to_delete_rtcp) {
     DestroyTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   }
 }
 }  // namespace webrtc