modules: more interface -> include renames

webrtc/modules/audio_coding/neteq/rtcp.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/rtcp.h"
 
 #include <string.h>
 
 #include <algorithm>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 void Rtcp::Init(uint16_t start_sequence_number) {
   cycles_ = 0;
   max_seq_no_ = start_sequence_number;
   base_seq_no_ = start_sequence_number;
   received_packets_ = 0;
   received_packets_prior_ = 0;
   expected_prior_ = 0;
@@ skipping 58 matching lines @@
   if (expected_since_last == 0 || lost <= 0 || received_packets_ == 0) {
     stats->fraction_lost = 0;
   } else {
     stats->fraction_lost = std::min(0xFFU, (lost << 8) / expected_since_last);
   }
 
   stats->jitter = jitter_ >> 4;  // Scaling from Q4.
 }
 
 }  // namespace webrtc