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

webrtc/modules/audio_processing/aec3/aec_state.cc

 /*
  *  Copyright (c) 2017 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/aec3/aec_state.h"
 
 #include <math.h>
 #include <numeric>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
-#include "webrtc/rtc_base/array_view.h"
-#include "webrtc/rtc_base/atomicops.h"
-#include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 namespace {
 
 constexpr size_t kEchoPathChangeConvergenceBlocks = 2 * kNumBlocksPerSecond;
 constexpr size_t kSaturationLeakageBlocks = 20;
 
 // Computes delay of the adaptive filter.
 rtc::Optional<size_t> EstimateFilterDelay(
     const std::vector<std::array<float, kFftLengthBy2Plus1>>&
@@ skipping 144 matching lines @@
   // After an amount of active render samples for which an echo should have been
   // detected in the capture signal if the ERL was not infinite, flag that a
   // headset is used.
   headset_detected_ =
       !external_delay_ && !filter_delay_ &&
       (!render_received_ ||
        blocks_with_filter_adaptation_ >= kEchoPathChangeConvergenceBlocks);
 }
 
 }  // namespace webrtc