Introduce struct MediaConfig, with construction-time settings.

webrtc/media/base/mediaengine.h

 /*
  *  Copyright (c) 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 44 matching lines @@
   // Initialization
   // Starts the engine.
   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;
+  virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
+                                           const MediaConfig& config,
+                                           const AudioOptions& options) = 0;
   // Creates a video media channel, paired with the specified voice channel.
   // Returns NULL on failure.
   virtual VideoMediaChannel* CreateVideoChannel(
       webrtc::Call* call,
+      const MediaConfig& config,
       const VideoOptions& 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 50 matching lines @@
     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 MediaConfig& config,
                                            const AudioOptions& options) {
-    return voice_.CreateChannel(call, options);
+    return voice_.CreateChannel(call, config, options);
   }
   virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
+                                                const MediaConfig& config,
                                                 const VideoOptions& options) {
-    return video_.CreateChannel(call, options);
+    return video_.CreateChannel(call, config, 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_