Removed GetOutputVolume() and SetOutputVolume().

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 54 matching lines @@
   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;
 
   virtual const std::vector<AudioCodec>& audio_codecs() = 0;
   virtual RtpCapabilities GetAudioCapabilities() = 0;
   virtual const std::vector<VideoCodec>& video_codecs() = 0;
   virtual RtpCapabilities GetVideoCapabilities() = 0;
 
   // Starts AEC dump using existing file, a maximum file size in bytes can be
   // specified. Logging is stopped just before the size limit is exceeded.
@@ skipping 55 matching lines @@
                                            const MediaConfig& config,
                                            const AudioOptions& options) {
     return voice_.CreateChannel(call, config, options);
   }
   virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
                                                 const MediaConfig& config,
                                                 const VideoOptions& 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() {
     return voice_.GetInputLevel();
   }
   virtual const std::vector<AudioCodec>& audio_codecs() {
     return voice_.codecs();
   }
   virtual RtpCapabilities GetAudioCapabilities() {
     return voice_.GetCapabilities();
   }
   virtual const std::vector<VideoCodec>& video_codecs() {
@@ skipping 34 matching lines @@
   virtual ~DataEngineInterface() {}
   virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
   virtual const std::vector<DataCodec>& data_codecs() = 0;
 };
 
 webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIAENGINE_H_