Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/pc/test/fakeaudiocapturemodule.cc

 /*
  *  Copyright 2012 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/pc/test/fakeaudiocapturemodule.h"
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/timeutils.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/thread.h"
+#include "webrtc/rtc_base/timeutils.h"
 
 // Audio sample value that is high enough that it doesn't occur naturally when
 // frames are being faked. E.g. NetEq will not generate this large sample value
 // unless it has received an audio frame containing a sample of this value.
 // Even simpler buffers would likely just contain audio sample values of 0.
 static const int kHighSampleValue = 10000;
 
 // Same value as src/modules/audio_device/main/source/audio_device_config.h in
 // https://code.google.com/p/webrtc/
 static const int kAdmMaxIdleTimeProcess = 1000;
@@ skipping 688 matching lines @@
                                               kNumberBytesPerSample,
                                               kNumberOfChannels,
                                               kSamplesPerSecond, kTotalDelayMs,
                                               kClockDriftMs, current_mic_level,
                                               key_pressed,
                                               current_mic_level) != 0) {
     RTC_NOTREACHED();
   }
   SetMicrophoneVolume(current_mic_level);
 }