Support for two audio codec lists down into WebRtcVoiceEngine.

webrtc/media/engine/webrtcvoiceengine.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 42 matching lines @@
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
                                    const MediaConfig& config,
                                    const AudioOptions& options);
 
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
   int GetInputLevel();
 
-  const std::vector<AudioCodec>& codecs();
+  const std::vector<AudioCodec>& send_codecs() const;
+  const std::vector<AudioCodec>& recv_codecs() const;
   RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel* channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
 
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
@@ skipping 212 matching lines @@
     int cng_payload_type = -1;
     int cng_plfreq = -1;
     webrtc::CodecInst codec_inst;
   } send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_