Delete all use of tick_util.h.

Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.

BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1888593004 .

Cr-Commit-Position: refs/heads/master@{#12674}
diff --git a/webrtc/modules/audio_coding/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 0507691..46c398b 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 @@
       _lastFrameSizeSample(0),
       _packetLoss(0),
       _useFECTestWithPacketLoss(false),
-      _beginTime(TickTime::MillisecondTimestamp()),
+      _beginTime(rtc::TimeMillis()),
       _totalBytes(0),
       external_send_timestamp_(-1),
       external_sequence_number_(-1),
@@ -286,7 +286,7 @@
       _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
     }
   }
-  _beginTime = TickTime::MillisecondTimestamp();
+  _beginTime = rtc::TimeMillis();
   _totalBytes = 0;
   _channelCritSect.Leave();
 }
@@ -411,7 +411,7 @@
 
 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();