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

webrtc/modules/audio_device/android/audio_device_unittest.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 <algorithm>
 #include <limits>
 #include <list>
 #include <memory>
 #include <numeric>
 #include <string>
 #include <vector>
 
+#include "webrtc/base/arraysize.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/build_info.h"
 #include "webrtc/modules/audio_device/android/ensure_initialized.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/arraysize.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using std::cout;
 using std::endl;
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::Gt;
@@ skipping 972 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc