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

webrtc/modules/audio_processing/aec3/block_framer.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_processing/aec3/block_framer.h"
 
 #include <algorithm>
 
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 
 BlockFramer::BlockFramer(size_t num_bands)
     : num_bands_(num_bands),
       buffer_(num_bands_, std::vector<float>(kBlockSize, 0.f)) {}
 
 BlockFramer::~BlockFramer() = default;
 
 // All the constants are chosen so that the buffer is either empty or has enough
@@ skipping 24 matching lines @@
     std::copy(buffer_[i].begin(), buffer_[i].end(), (*sub_frame)[i].begin());
     std::copy(block[i].begin(), block[i].begin() + samples_to_frame,
               (*sub_frame)[i].begin() + buffer_[i].size());
     buffer_[i].clear();
     buffer_[i].insert(buffer_[i].begin(), block[i].begin() + samples_to_frame,
                       block[i].end());
   }
 }
 
 }  // namespace webrtc