system_wrappers: rename interface -> include

webrtc/modules/audio_coding/neteq/decision_logic.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.h"
 
 #include <algorithm>
 
 #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
 #include "webrtc/modules/audio_coding/neteq/decision_logic_normal.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/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
 DecisionLogic* DecisionLogic::Create(int fs_hz,
                                      size_t output_size_samples,
                                      NetEqPlayoutMode playout_mode,
                                      DecoderDatabase* decoder_database,
                                      const PacketBuffer& packet_buffer,
                                      DelayManager* delay_manager,
                                      BufferLevelFilter* buffer_level_filter) {
@@ skipping 144 matching lines @@
     }
     buffer_level_filter_->Update(buffer_size_packets, sample_memory_local,
                                  packet_length_samples_);
     prev_time_scale_ = false;
   }
 
   timescale_hold_off_ = std::max(timescale_hold_off_ - 1, 0);
 }
 
 }  // namespace webrtc