Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

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 <cmath>
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <cstdio>
 #endif
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/safe_minmax.h"
 #include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
 #include "webrtc/modules/audio_processing/gain_control_impl.h"
 #include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_minmax.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
 namespace {
 
 // Amount the microphone level is lowered with every clipping event.
 const int kClippedLevelStep = 15;
 // Proportion of clipped samples required to declare a clipping event.
 const float kClippedRatioThreshold = 0.1f;
@@ skipping 413 matching lines @@
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
       LOG(LS_ERROR) << "set_compression_gain_db(" << compression_
                     << ") failed.";
     }
   }
 }
 
 }  // namespace webrtc