modules: more interface -> include renames

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

 /*
  *  Copyright (c) 2012 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_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
 
 #include <stddef.h>
 #include <list>
 
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 #define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
 #define IP_PACKET_SIZE 1500    // we assume ethernet
 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
 #define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
 
 namespace webrtc {
 namespace rtcp {
@@ skipping 402 matching lines @@
 
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_