Support for two audio codec lists down into WebRtcVoiceEngine.

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 62 matching lines @@
 
   // 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 const std::vector<AudioCodec>& audio_send_codecs() = 0;
+  virtual const std::vector<AudioCodec>& audio_recv_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.
   // If max_size_bytes is set to a value <= 0, no limit will be used.
   virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
 
   // Stops recording AEC dump.
@@ skipping 54 matching lines @@
   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 const std::vector<AudioCodec>& audio_send_codecs() {
+    return voice_.send_codecs();
+  }
+  virtual const std::vector<AudioCodec>& audio_recv_codecs() {
+    return voice_.recv_codecs();
   }
   virtual RtpCapabilities GetAudioCapabilities() {
     return voice_.GetCapabilities();
   }
   virtual const std::vector<VideoCodec>& video_codecs() {
     return video_.codecs();
   }
   virtual RtpCapabilities GetVideoCapabilities() {
     return video_.GetCapabilities();
   }
@@ skipping 28 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_