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

webrtc/modules/audio_processing/gain_control_for_experimental_agc.h

 /*
  *  Copyright (c) 2016 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
 
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/thread_checker.h"
 
 namespace webrtc {
 
 class ApmDataDumper;
 
 // This class has two main purposes:
 //
 // 1) It is returned instead of the real GainControl after the new AGC has been
 //    enabled in order to prevent an outside user from overriding compression
 //    settings. It doesn't do anything in its implementation, except for
@@ skipping 39 matching lines @@
   GainControl* real_gain_control_;
   int volume_;
   rtc::CriticalSection* crit_capture_;
   static int instance_counter_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_