Define Stream base classes

webrtc/call.h

 /*
  *  Copyright (c) 2013 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_CALL_H_
@@ skipping 50 matching lines @@
 
  protected:
   virtual ~LoadObserver() {}
 };
 
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
 class Call {
  public:
-  enum NetworkState {
-    kNetworkUp,
-    kNetworkDown,
-  };
   struct Config {
     Config() = delete;
     explicit Config(newapi::Transport* send_transport)
         : send_transport(send_transport) {}
 
     static const int kDefaultStartBitrateBps;
 
     // TODO(solenberg): Need to add media type to the interface for outgoing
     // packets too.
     newapi::Transport* send_transport = nullptr;
@@ skipping 61 matching lines @@
   // of maximum for entire Call. This should be fixed along with the above.
   // Specifying a start bitrate (>0) will currently reset the current bitrate
   // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
   virtual ~Call() {}
 };
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_