Delete MediaController class, move Call ownership to PeerConnection.

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.
  */
 
 #ifndef WEBRTC_PC_CHANNELMANAGER_H_
 #define WEBRTC_PC_CHANNELMANAGER_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/mediaengine.h"
 #include "webrtc/pc/voicechannel.h"
 
-namespace webrtc {
-class MediaControllerInterface;
-}
 namespace cricket {
 
 class VoiceChannel;
 
 // ChannelManager allows the MediaEngine to run on a separate thread, and takes
 // care of marshalling calls between threads. It also creates and keeps track of
 // 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.
@@ skipping 45 matching lines @@
   // Indicates whether the media engine is started.
   bool initialized() const { return initialized_; }
   // Starts up the media engine.
   bool Init();
   // Shuts down the media engine.
   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,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      const cricket::MediaConfig& media_config,
       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.
   void DestroyRtpDataChannel(RtpDataChannel* data_channel);
 
   // Indicates whether any channels exist.
   bool has_channels() const {
@@ skipping 23 matching lines @@
   typedef std::vector<RtpDataChannel*> RtpDataChannels;
 
   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();
   VoiceChannel* CreateVoiceChannel_w(
-      webrtc::MediaControllerInterface* media_controller,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
       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,
+      const cricket::MediaConfig& media_config,
       DtlsTransportInternal* rtp_transport,
       DtlsTransportInternal* rtcp_transport,
       rtc::Thread* signaling_thread,
       const std::string& content_name,
       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_