Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

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 78 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,
       DtlsTransportInternal* rtp_transport,
       DtlsTransportInternal* rtcp_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
       const AudioOptions& options);
+  // Version of the above that takes PacketTransportInternal.
+  VoiceChannel* CreateVoiceChannel(
+      webrtc::MediaControllerInterface* media_controller,
+      rtc::PacketTransportInternal* rtp_transport,
+      rtc::PacketTransportInternal* rtcp_transport,
+      rtc::Thread* signaling_thread,
+      const std::string& content_name,
+      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,
       DtlsTransportInternal* rtp_transport,
       DtlsTransportInternal* rtcp_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
       const VideoOptions& options);
+  // Version of the above that takes PacketTransportInternal.
+  VideoChannel* CreateVideoChannel(
+      webrtc::MediaControllerInterface* media_controller,
+      rtc::PacketTransportInternal* rtp_transport,
+      rtc::PacketTransportInternal* rtcp_transport,
+      rtc::Thread* signaling_thread,
+      const std::string& content_name,
+      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,
       DtlsTransportInternal* rtp_transport,
       DtlsTransportInternal* rtcp_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required);
   // Destroys a data channel created with the Create API.
@@ skipping 33 matching lines @@
   void Construct(std::unique_ptr<MediaEngineInterface> me,
                  std::unique_ptr<DataEngineInterface> dme,
                  rtc::Thread* worker_thread,
                  rtc::Thread* network_thread);
   bool InitMediaEngine_w();
   void DestructorDeletes_w();
   void Terminate_w();
   bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options);
   VoiceChannel* CreateVoiceChannel_w(
       webrtc::MediaControllerInterface* media_controller,
-      DtlsTransportInternal* rtp_transport,
-      DtlsTransportInternal* rtcp_transport,
+      DtlsTransportInternal* rtp_dtls_transport,
+      DtlsTransportInternal* rtcp_dtls_transport,
+      rtc::PacketTransportInternal* rtp_packet_transport,
+      rtc::PacketTransportInternal* rtcp_packet_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
       const AudioOptions& options);
   void DestroyVoiceChannel_w(VoiceChannel* voice_channel);
   VideoChannel* CreateVideoChannel_w(
       webrtc::MediaControllerInterface* media_controller,
-      DtlsTransportInternal* rtp_transport,
-      DtlsTransportInternal* rtcp_transport,
+      DtlsTransportInternal* rtp_dtls_transport,
+      DtlsTransportInternal* rtcp_dtls_transport,
+      rtc::PacketTransportInternal* rtp_packet_transport,
+      rtc::PacketTransportInternal* rtcp_packet_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       bool srtp_required,
       const VideoOptions& options);
   void DestroyVideoChannel_w(VideoChannel* video_channel);
   RtpDataChannel* CreateRtpDataChannel_w(
       webrtc::MediaControllerInterface* media_controller,
       DtlsTransportInternal* rtp_transport,
       DtlsTransportInternal* rtcp_transport,
       rtc::Thread* signaling_thread,
@@ skipping 14 matching lines @@
 
   bool enable_rtx_;
   rtc::CryptoOptions crypto_options_;
 
   bool capturing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_PC_CHANNELMANAGER_H_