Move webrtc/audio_*.h to webrtc/api/call

webrtc/api/call/audio_send_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_AUDIO_SEND_STREAM_H_
-#define WEBRTC_AUDIO_SEND_STREAM_H_
+#ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
+#define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ skipping 77 matching lines @@
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_SEND_STREAM_H_
+#endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_