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

webrtc/modules/audio_processing/gain_control_impl.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
 
 #include <memory>
 #include <vector>
 
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/swap_queue.h"
-#include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/swap_queue.h"
+#include "webrtc/rtc_base/thread_annotations.h"
 
 namespace webrtc {
 
 class ApmDataDumper;
 class AudioBuffer;
 
 class GainControlImpl : public GainControl {
  public:
   GainControlImpl(rtc::CriticalSection* crit_render,
                   rtc::CriticalSection* crit_capture);
@@ skipping 55 matching lines @@
 
   rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
   rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
 
   static int instance_counter_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_