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

webrtc/modules/audio_device/android/audio_manager_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 <SLES/OpenSLES_Android.h>
 
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/format_macros.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/rtc_base/arraysize.h"
+#include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/test/gtest.h"
 
 #define PRINT(...) fprintf(stderr, __VA_ARGS__);
 
 namespace webrtc {
 
 static const char kTag[] = "  ";
 
 class AudioManagerTest : public ::testing::Test {
  protected:
@@ skipping 194 matching lines @@
             params.frames_per_10ms_buffer());
   EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
   EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
   EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
             params.GetBytesPer10msBuffer());
   EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
 }
 
 }  // namespace webrtc