system_wrappers: rename interface -> include

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  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.
  */
 
@@ skipping 24 matching lines @@
 #include "webrtc/modules/audio_coding/neteq/merge.h"
 #include "webrtc/modules/audio_coding/neteq/normal.h"
 #include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/modules/audio_coding/neteq/payload_splitter.h"
 #include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
 #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
 #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
 #include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 
 // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
 // longer required, this #define should be removed (and the code that it
 // enables).
 #define LEGACY_BITEXACT
 
 namespace webrtc {
 
 NetEqImpl::NetEqImpl(const NetEq::Config& config,
                      BufferLevelFilter* buffer_level_filter,
@@ skipping 1970 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc