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

webrtc/modules/audio_device/audio_device_unittest.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 <algorithm>
 #include <cstring>
 #include <numeric>
 
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/race_checker.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
-#include "webrtc/rtc_base/array_view.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/event.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/optional.h"
-#include "webrtc/rtc_base/race_checker.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/rtc_base/thread_checker.h"
-#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::Ge;
 using ::testing::Invoke;
 using ::testing::NiceMock;
 using ::testing::NotNull;
 
@@ skipping 650 matching lines @@
   StopPlayout();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(audio_stream.num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   // Print out min, max and average delay values for debugging purposes.
   audio_stream.PrintResults();
 }
 
 }  // namespace webrtc