modules: more interface -> include renames

webrtc/modules/audio_processing/agc/agc_manager_direct.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_processing/agc/agc_manager_direct.h"
 
 #include <cassert>
 #include <cmath>
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <cstdio>
 #endif
 
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 namespace webrtc {
 
 namespace {
 
 // Lowest the microphone level can be lowered due to clipping.
 const int kClippedLevelMin = 170;
 // Amount the microphone level is lowered with every clipping event.
 const int kClippedLevelStep = 15;
@@ skipping 400 matching lines @@
   if (new_compression != compression_) {
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
       LOG_FERR1(LS_ERROR, set_compression_gain_db, compression_);
     }
   }
 }
 
 }  // namespace webrtc