Delete all use of tick_util.h.

webrtc/modules/audio_coding/test/APITest.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/APITest.h"
 
 #include <ctype.h>
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include <iostream>
 #include <ostream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/common_types.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 #define TEST_DURATION_SEC 600
 #define NUMBER_OF_SENDER_TESTS 6
 #define MAX_FILE_NAME_LENGTH_BYTE 500
 
 void APITest::Wait(uint32_t waitLengthMs) {
@@ skipping 512 matching lines @@
 
   _pullEventA->StartTimer(true, 10);
   _pullEventB->StartTimer(true, 10);
 
   _pushEventA->StartTimer(true, 10);
   _pushEventB->StartTimer(true, 10);
 
   // Keep main thread waiting for sender/receiver
   // threads to complete
   EventWrapper* completeEvent = EventWrapper::Create();
-  uint64_t startTime = TickTime::MillisecondTimestamp();
+  uint64_t startTime = rtc::TimeMillis();
   uint64_t currentTime;
   // Run test in 2 minutes (120000 ms).
   do {
     {
       //ReadLockScoped rl(_apiTestRWLock);
       //fprintf(stderr, "\r%s", _movingDot);
     }
     //fflush(stderr);
     completeEvent->Wait(50);
-    currentTime = TickTime::MillisecondTimestamp();
+    currentTime = rtc::TimeMillis();
   } while ((currentTime - startTime) < 120000);
 
   //completeEvent->Wait(0xFFFFFFFF);
   //(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
   delete completeEvent;
 
   myPushAudioThreadA.Stop();
   myPullAudioThreadA.Stop();
   myProcessThreadA.Stop();
   myAPIThreadA.Stop();
@@ skipping 512 matching lines @@
   CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
   myChannel->ResetStats();
   {
     WriteLockScoped wl(_apiTestRWLock);
     *thereIsEncoder = true;
   }
   Wait(500);
 }
 
 }  // namespace webrtc