Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/media/engine/webrtcmediaengine.h

 /*
  *  Copyright (c) 2011 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.
  */
 
 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/call/call.h"
-#include "webrtc/config.h"
 #include "webrtc/media/base/mediaengine.h"
 
 namespace webrtc {
 class AudioDecoderFactory;
 class AudioDeviceModule;
 class AudioMixer;
 class AudioProcessing;
 }
 namespace cricket {
 class WebRtcVideoDecoderFactory;
@@ skipping 66 matching lines @@
     const std::vector<webrtc::RtpExtension>& extensions,
     bool (*supported)(const std::string&),
     bool filter_redundant_extensions);
 
 webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
     const Codec& codec);
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCMEDIAENGINE_H_