Delete all use of tick_util.h.

webrtc/modules/audio_coding/test/Channel.cc

Index: webrtc/modules/audio_coding/test/Channel.cc
diff --git a/webrtc/modules/audio_coding/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 0507691fb4dcdbb8ef31d2bf21376ec4c684c8ea..ff984a4c91bc066c61f759c203d0ed3c32eebf08 100644
--- a/webrtc/modules/audio_coding/test/Channel.cc
+++ b/webrtc/modules/audio_coding/test/Channel.cc
@@ -14,7 +14,7 @@
 #include <iostream>
 
 #include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/base/timeutils.h"
 
 namespace webrtc {
 
@@ -234,7 +234,7 @@ Channel::Channel(int16_t chID)
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
-      _beginTime(TickTime::MillisecondTimestamp()),
+      _beginTime(rtc::Time64()),
       _totalBytes(0),
       external_send_timestamp_(-1),
       external_sequence_number_(-1),
@@ -286,7 +286,7 @@ void Channel::ResetStats() {
       _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
     }
   }
-  _beginTime = TickTime::MillisecondTimestamp();
+  _beginTime = rtc::Time64();
   _totalBytes = 0;
   _channelCritSect.Leave();
 }
@@ -411,7 +411,7 @@ uint32_t Channel::LastInTimestamp() {
 
 double Channel::BitRate() {
   double rate;
-  uint64_t currTime = TickTime::MillisecondTimestamp();
+  uint64_t currTime = rtc::Time64();
   _channelCritSect.Enter();
   rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
   _channelCritSect.Leave();