modules: more interface -> include renames

webrtc/modules/audio_coding/neteq/decision_logic_normal.cc

 /*
  *  Copyright (c) 2013 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/decision_logic_normal.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
 #include "webrtc/modules/audio_coding/neteq/decoder_database.h"
 #include "webrtc/modules/audio_coding/neteq/delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/expand.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 Operations DecisionLogicNormal::GetDecisionSpecialized(
     const SyncBuffer& sync_buffer,
     const Expand& expand,
     size_t decoder_frame_length,
     const RTPHeader* packet_header,
     Modes prev_mode,
     bool play_dtmf,
@@ skipping 195 matching lines @@
 bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
   return timestamp_leap >
       static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
 }
 
 bool DecisionLogicNormal::MaxWaitForPacket() const {
   return num_consecutive_expands_ >= kMaxWaitForPacket;
 }
 
 }  // namespace webrtc