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

webrtc/modules/audio_processing/test/debug_dump_replayer.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/test/debug_dump_replayer.h"
 
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
-
+#include "webrtc/rtc_base/checks.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 
 void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
                       const StreamConfig& config) {
   auto& buffer_ref = *buffer;
   if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
@@ skipping 238 matching lines @@
   apm_->ApplyConfig(apm_config);
 }
 
 void DebugDumpReplayer::LoadNextMessage() {
   has_next_event_ =
       debug_file_ && ReadMessageFromFile(debug_file_, &next_event_);
 }
 
 }  // namespace test
 }  // namespace webrtc