Delete all use of tick_util.h.

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/system_wrappers/include/tick_util.h"
+#include "webrtc/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 199 matching lines @@
       _bitStreamFile(NULL),
       _saveBitStream(false),
       _lastPayloadType(-1),
       _isStereo(false),
       _leftChannel(true),
       _lastInTimestamp(0),
       _useLastFrameSize(false),
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
-      _beginTime(TickTime::MillisecondTimestamp()),
+      _beginTime(rtc::TimeMillis()),
       _totalBytes(0),
       external_send_timestamp_(-1),
       external_sequence_number_(-1),
       num_packets_to_drop_(0) {
   int n;
   int k;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
     _payloadStats[n].payloadType = -1;
     _payloadStats[n].newPacket = true;
     for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
@@ skipping 31 matching lines @@
     _payloadStats[n].payloadType = -1;
     _payloadStats[n].newPacket = true;
     for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
       _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
       _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
       _payloadStats[n].frameSizeStats[k].numPackets = 0;
       _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
       _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
     }
   }
-  _beginTime = TickTime::MillisecondTimestamp();
+  _beginTime = rtc::TimeMillis();
   _totalBytes = 0;
   _channelCritSect.Leave();
 }
 
 int16_t Channel::Stats(CodecInst& codecInst,
                        ACMTestPayloadStats& payloadStats) {
   _channelCritSect.Enter();
   int n;
   payloadStats.payloadType = -1;
   for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
@@ skipping 104 matching lines @@
 uint32_t Channel::LastInTimestamp() {
   uint32_t timestamp;
   _channelCritSect.Enter();
   timestamp = _lastInTimestamp;
   _channelCritSect.Leave();
   return timestamp;
 }
 
 double Channel::BitRate() {
   double rate;
-  uint64_t currTime = TickTime::MillisecondTimestamp();
+  uint64_t currTime = rtc::TimeMillis();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();
   return rate;
 }
 
 }  // namespace webrtc