Refactoring of RTCP options in BaseChannel.

webrtc/pc/channelmanager.cc

 /*
  *  Copyright 2004 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 194 matching lines @@
   }
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required,
     const AudioOptions& options) {
   return worker_thread_->Invoke<VoiceChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateVoiceChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
-           bundle_transport_name, rtcp, srtp_required, options));
+           bundle_transport_name, rtcp_mux_required, srtp_required, options));
 }
 
 VoiceChannel* ChannelManager::CreateVoiceChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required,
     const AudioOptions& options) {
   RTC_DCHECK(initialized_);
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   RTC_DCHECK(nullptr != media_controller);
 
   VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (!media_channel)
     return nullptr;
 
   VoiceChannel* voice_channel = new VoiceChannel(
       worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
-      media_channel, content_name, rtcp, srtp_required);
+      media_channel, content_name, rtcp_mux_required, srtp_required);
   voice_channel->SetCryptoOptions(crypto_options_);
 
   if (!voice_channel->Init_w(rtp_transport, rtcp_transport)) {
     delete voice_channel;
     return nullptr;
   }
   voice_channels_.push_back(voice_channel);
   return voice_channel;
 }
 
@@ skipping 20 matching lines @@
   delete voice_channel;
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required,
     const VideoOptions& options) {
   return worker_thread_->Invoke<VideoChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateVideoChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
-           bundle_transport_name, rtcp, srtp_required, options));
+           bundle_transport_name, rtcp_mux_required, srtp_required, options));
 }
 
 VideoChannel* ChannelManager::CreateVideoChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required,
     const VideoOptions& options) {
   RTC_DCHECK(initialized_);
   RTC_DCHECK(worker_thread_ == rtc::Thread::Current());
   RTC_DCHECK(nullptr != media_controller);
   VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
       media_controller->call_w(), media_controller->config(), options);
   if (media_channel == NULL) {
     return NULL;
   }
 
-  VideoChannel* video_channel =
-      new VideoChannel(worker_thread_, network_thread_, signaling_thread,
-                       media_channel, content_name, rtcp, srtp_required);
+  VideoChannel* video_channel = new VideoChannel(
+      worker_thread_, network_thread_, signaling_thread, media_channel,
+      content_name, rtcp_mux_required, srtp_required);
   video_channel->SetCryptoOptions(crypto_options_);
   if (!video_channel->Init_w(rtp_transport, rtcp_transport)) {
     delete video_channel;
     return NULL;
   }
   video_channels_.push_back(video_channel);
   return video_channel;
 }
 
 void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
@@ skipping 20 matching lines @@
   delete video_channel;
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required) {
   return worker_thread_->Invoke<RtpDataChannel*>(
       RTC_FROM_HERE,
       Bind(&ChannelManager::CreateRtpDataChannel_w, this, media_controller,
            rtp_transport, rtcp_transport, signaling_thread, content_name,
-           bundle_transport_name, rtcp, srtp_required));
+           bundle_transport_name, rtcp_mux_required, srtp_required));
 }
 
 RtpDataChannel* ChannelManager::CreateRtpDataChannel_w(
     webrtc::MediaControllerInterface* media_controller,
     TransportChannel* rtp_transport,
     TransportChannel* rtcp_transport,
     rtc::Thread* signaling_thread,
     const std::string& content_name,
     const std::string* bundle_transport_name,
-    bool rtcp,
+    bool rtcp_mux_required,
     bool srtp_required) {
   // This is ok to alloc from a thread other than the worker thread.
   RTC_DCHECK(initialized_);
   MediaConfig config;
   if (media_controller) {
     config = media_controller->config();
   }
   DataMediaChannel* media_channel = data_media_engine_->CreateChannel(config);
   if (!media_channel) {
     LOG(LS_WARNING) << "Failed to create RTP data channel.";
     return nullptr;
   }
 
-  RtpDataChannel* data_channel =
-      new RtpDataChannel(worker_thread_, network_thread_, signaling_thread,
-                         media_channel, content_name, rtcp, srtp_required);
+  RtpDataChannel* data_channel = new RtpDataChannel(
+      worker_thread_, network_thread_, signaling_thread, media_channel,
+      content_name, rtcp_mux_required, srtp_required);
   data_channel->SetCryptoOptions(crypto_options_);
   if (!data_channel->Init_w(rtp_transport, rtcp_transport)) {
     LOG(LS_WARNING) << "Failed to init data channel.";
     delete data_channel;
     return nullptr;
   }
   data_channels_.push_back(data_channel);
   return data_channel;
 }
 
@@ skipping 27 matching lines @@
                           media_engine_.get(), file, max_size_bytes));
 }
 
 void ChannelManager::StopAecDump() {
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
       Bind(&MediaEngineInterface::StopAecDump, media_engine_.get()));
 }
 
 }  // namespace cricket