Delete MediaController class, move Call ownership to PeerConnection.

webrtc/ortc/rtptransportcontrolleradapter.h

 /*
  *  Copyright 2017 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_ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
 #define WEBRTC_ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
 #include "webrtc/api/ortc/ortcrtpsenderinterface.h"
 #include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
 #include "webrtc/api/ortc/srtptransportinterface.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/mediachannel.h"  // For MediaConfig.
 #include "webrtc/pc/channelmanager.h"
-#include "webrtc/pc/mediacontroller.h"
 
 namespace webrtc {
 
 class RtpTransportAdapter;
 class OrtcRtpSenderAdapter;
 class OrtcRtpReceiverAdapter;
 
 // Implementation of RtpTransportControllerInterface. Wraps a MediaController,
 // a VoiceChannel and VideoChannel, and maintains a list of dependent RTP
 // transports.
@@ skipping 82 matching lines @@
  protected:
   RtpTransportControllerAdapter* GetInternal() override { return this; }
 
  private:
   // Only expected to be called by RtpTransportControllerAdapter::CreateProxied.
   RtpTransportControllerAdapter(const cricket::MediaConfig& config,
                                 cricket::ChannelManager* channel_manager,
                                 webrtc::RtcEventLog* event_log,
                                 rtc::Thread* signaling_thread,
                                 rtc::Thread* worker_thread);
+  void Init_w();
+  void Close_w();
 
   // These return an error if another of the same type of object is already
   // attached, or if |transport_proxy| can't be used with the sender/receiver
   // due to the limitation that the sender/receiver of the same media type must
   // use the same transport.
   RTCError AttachAudioSender(OrtcRtpSenderAdapter* sender,
                              RtpTransportInterface* inner_transport);
   RTCError AttachVideoSender(OrtcRtpSenderAdapter* sender,
                              RtpTransportInterface* inner_transport);
   RTCError AttachAudioReceiver(OrtcRtpReceiverAdapter* receiver,
@@ skipping 41 matching lines @@
       cricket::MediaContentDescription* remote_description);
 
   rtc::Thread* signaling_thread_;
   rtc::Thread* worker_thread_;
   // |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_|
   // are somewhat redundant, but the latter are only set when
   // RtpSenders/RtpReceivers are attached to the transport.
   std::vector<RtpTransportInterface*> transport_proxies_;
   RtpTransportInterface* inner_audio_transport_ = nullptr;
   RtpTransportInterface* inner_video_transport_ = nullptr;
-  std::unique_ptr<MediaControllerInterface> media_controller_;
+  const cricket::MediaConfig media_config_;
+  cricket::ChannelManager* channel_manager_;
+  webrtc::RtcEventLog* event_log_;
+  std::unique_ptr<Call> call_;
 
   // BaseChannel takes content descriptions as input, so we store them here
   // such that they can be updated when a new RtpSenderAdapter/
   // RtpReceiverAdapter attaches itself.
   cricket::AudioContentDescription local_audio_description_;
   cricket::AudioContentDescription remote_audio_description_;
   cricket::VideoContentDescription local_video_description_;
   cricket::VideoContentDescription remote_video_description_;
   cricket::VoiceChannel* voice_channel_ = nullptr;
   cricket::VideoChannel* video_channel_ = nullptr;
   bool have_audio_sender_ = false;
   bool have_video_sender_ = false;
   bool have_audio_receiver_ = false;
   bool have_video_receiver_ = false;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportControllerAdapter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_ORTC_RTPTRANSPORTCONTROLLERADAPTER_H_