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 21 matching lines @@
 // voice and video channels; by doing so, it can temporarily pause all the
 // channels when a new audio or video device is chosen. The voice and video
 // channels are stored in separate vectors, to easily allow operations on just
 // voice or just video channels.
 // ChannelManager also allows the application to discover what devices it has
 // using device manager.
 class ChannelManager {
  public:
   // For testing purposes. Allows the media engine and data media
   // engine and dev manager to be mocks.  The ChannelManager takes
-  // ownership of these objects.
-  ChannelManager(MediaEngineInterface* me,
-                 DataEngineInterface* dme,
+  // ownership of the objects passed as unique_ptrs.
+  ChannelManager(std::unique_ptr<MediaEngineInterface> me,
+                 std::unique_ptr<DataEngineInterface> dme,
                  rtc::Thread* worker_and_network);
   // Same as above, but gives an easier default DataEngine.
-  ChannelManager(MediaEngineInterface* me,
+  ChannelManager(std::unique_ptr<MediaEngineInterface> me,
                  rtc::Thread* worker,
                  rtc::Thread* network);
   ~ChannelManager();
 
   // Accessors for the worker thread, allowing it to be set after construction,
   // but before Init. set_worker_thread will return false if called after Init.
   rtc::Thread* worker_thread() const { return worker_thread_; }
   bool set_worker_thread(rtc::Thread* thread) {
     if (initialized_) {
       return false;
@@ skipping 29 matching lines @@
   void Terminate();
 
   // 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,
-      const std::string* bundle_transport_name,
-      bool rtcp_mux_required,
+      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,
-      const std::string* bundle_transport_name,
-      bool rtcp_mux_required,
+      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,
-      const std::string* bundle_transport_name,
-      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 15 matching lines @@
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops recording AEC dump.
   void StopAecDump();
 
  private:
   typedef std::vector<VoiceChannel*> VoiceChannels;
   typedef std::vector<VideoChannel*> VideoChannels;
   typedef std::vector<RtpDataChannel*> RtpDataChannels;
 
-  void Construct(MediaEngineInterface* me,
-                 DataEngineInterface* dme,
+  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,
-      const std::string* bundle_transport_name,
-      bool rtcp_mux_required,
       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,
-      const std::string* bundle_transport_name,
-      bool rtcp_mux_required,
       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,
       const std::string& content_name,
-      const std::string* bundle_transport_name,
-      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_