Moved call.h and most of api/call/* into call/

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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIAENGINE_H_
 #define WEBRTC_MEDIA_BASE_MEDIAENGINE_H_
 
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 #include <CoreAudio/CoreAudio.h>
 #endif
 
 #include <string>
 #include <vector>
 
-#include "webrtc/api/call/audio_state.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/sigslotrepeater.h"
+#include "webrtc/call/audio_state.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videocommon.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 
 #if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
 #define DISABLE_MEDIA_ENGINE_FACTORY
 #endif
 
 namespace webrtc {
@@ skipping 137 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_