Refactoring of RTCP options in BaseChannel.

webrtc/pc/channelmanager.h

 /*
  *  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 77 matching lines @@
 
   // The operations below all occur on the worker thread.
   // Creates a voice channel, to be associated with the specified session.
   VoiceChannel* 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);
   // Destroys a voice channel created with the Create API.
   void DestroyVoiceChannel(VoiceChannel* voice_channel);
   // Creates a video channel, synced with the specified voice channel, and
   // associated with the specified session.
   VideoChannel* 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);
   // Destroys a video channel created with the Create API.
   void DestroyVideoChannel(VideoChannel* video_channel);
   RtpDataChannel* 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);
   // Destroys a data channel created with the Create API.
   void DestroyRtpDataChannel(RtpDataChannel* data_channel);
 
   // Indicates whether any channels exist.
   bool has_channels() const {
     return (!voice_channels_.empty() || !video_channels_.empty());
   }
 
   // RTX will be enabled/disabled in engines that support it. The supporting
@@ skipping 30 matching lines @@
   void DestructorDeletes_w();
   void Terminate_w();
   bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options);
   VoiceChannel* 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);
   void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
   VideoChannel* 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);
   void DestroyVideoChannel_w(VideoChannel* video_channel);
   RtpDataChannel* 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);
   void DestroyRtpDataChannel_w(RtpDataChannel* data_channel);
 
   std::unique_ptr<MediaEngineInterface> media_engine_;
   std::unique_ptr<DataEngineInterface> data_media_engine_;
   bool initialized_;
   rtc::Thread* main_thread_;
   rtc::Thread* worker_thread_;
   rtc::Thread* network_thread_;
 
   VoiceChannels voice_channels_;
   VideoChannels video_channels_;
   RtpDataChannels data_channels_;
 
   bool enable_rtx_;
   rtc::CryptoOptions crypto_options_;
 
   bool capturing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNELMANAGER_H_