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

webrtc/modules/audio_processing/test/debug_dump_test.cc

 /*
  *  Copyright (c) 2015 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 <stddef.h>  // size_t
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/task_queue.h"
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/test/debug_dump_replayer.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.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() ||
       buffer_ref->num_channels() != config.num_channels()) {
@@ skipping 556 matching lines @@
   config.Set<ExperimentalNs>(new ExperimentalNs(true));
   DebugDumpGenerator generator(config, AudioProcessing::Config());
   generator.StartRecording();
   generator.Process(100);
   generator.StopRecording();
   VerifyDebugDump(generator.dump_file_name());
 }
 
 }  // namespace test
 }  // namespace webrtc