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

webrtc/modules/audio_coding/test/Channel.cc

 /*
  *  Copyright (c) 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/modules/audio_coding/test/Channel.h"
 
 #include <assert.h>
 #include <iostream>
 
-#include "webrtc/base/format_macros.h"
-#include "webrtc/base/timeutils.h"
+#include "webrtc/rtc_base/format_macros.h"
+#include "webrtc/rtc_base/timeutils.h"
 
 namespace webrtc {
 
 int32_t Channel::SendData(FrameType frameType,
                           uint8_t payloadType,
                           uint32_t timeStamp,
                           const uint8_t* payloadData,
                           size_t payloadSize,
                           const RTPFragmentationHeader* fragmentation) {
   WebRtcRTPHeader rtpInfo;
@@ skipping 384 matching lines @@
 double Channel::BitRate() {
   double rate;
   uint64_t currTime = rtc::TimeMillis();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();
   return rate;
 }
 
 }  // namespace webrtc