Moved transport.h from webrtc/ to webrtc/api, created build target and updated dependencies.

webrtc/api/call/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 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_API_CALL_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
+#include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
-#include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class AudioSinkInterface;
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
@@ skipping 98 matching lines @@
   // Sets playback gain of the stream, applied when mixing, and thus after it
   // is potentially forwarded to any attached AudioSinkInterface implementation.
   virtual void SetGain(float gain) = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_