Split out dscp option from VideoOptions to new struct MediaChannelOptions.

webrtc/media/base/mediaengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 63 matching lines @@
   virtual bool Init(rtc::Thread* worker_thread) = 0;
   // Shuts down the engine.
   virtual void Terminate() = 0;
   // TODO(solenberg): Remove once VoE API refactoring is done.
   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
 
   // MediaChannel creation
   // Creates a voice media channel. Returns NULL on failure.
   virtual VoiceMediaChannel* CreateChannel(
       webrtc::Call* call,
-      const AudioOptions& options) = 0;
+      const MediaChannelOptions& options,
+      const AudioOptions& audio_options) = 0;
   // Creates a video media channel, paired with the specified voice channel.
   // Returns NULL on failure.
   virtual VideoMediaChannel* CreateVideoChannel(
       webrtc::Call* call,
-      const VideoOptions& options) = 0;
+      const MediaChannelOptions& options,
+      const VideoOptions& video_options) = 0;
 
   // Device configuration
   // Gets the current speaker volume, as a value between 0 and 255.
   virtual bool GetOutputVolume(int* level) = 0;
   // Sets the current speaker volume, as a value between 0 and 255.
   virtual bool SetOutputVolume(int level) = 0;
 
   // Gets the current microphone level, as a value between 0 and 10.
   virtual int GetInputLevel() = 0;
 
@@ skipping 48 matching lines @@
     video_.Init();
     return true;
   }
   virtual void Terminate() {
     voice_.Terminate();
   }
 
   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
     return voice_.GetAudioState();
   }
-  virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
-                                           const AudioOptions& options) {
-    return voice_.CreateChannel(call, options);
+  virtual VoiceMediaChannel* CreateChannel(
+      webrtc::Call* call,
+      const MediaChannelOptions& options,
+      const AudioOptions& audio_options) {
+    return voice_.CreateChannel(call, options, audio_options);
   }
-  virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
-                                                const VideoOptions& options) {
-    return video_.CreateChannel(call, options);
+  virtual VideoMediaChannel* CreateVideoChannel(
+      webrtc::Call* call,
+      const MediaChannelOptions& options,
+      const VideoOptions& video_options) {
+    return video_.CreateChannel(call, options, video_options);
   }
 
   virtual bool GetOutputVolume(int* level) {
     return voice_.GetOutputVolume(level);
   }
   virtual bool SetOutputVolume(int level) {
     return voice_.SetOutputVolume(level);
   }
 
   virtual int GetInputLevel() {
@@ skipping 40 matching lines @@
 class DataEngineInterface {
  public:
   virtual ~DataEngineInterface() {}
   virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
   virtual const std::vector<DataCodec>& data_codecs() = 0;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIAENGINE_H_