Delete Timing class, timing.h, and update all users.

webrtc/stats/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 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/stats/rtcstatscollector.h"
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/rtcstats_objects.h"
 #include "webrtc/api/rtcstatsreport.h"
 #include "webrtc/api/test/mock_datachannel.h"
 #include "webrtc/api/test/mock_peerconnection.h"
 #include "webrtc/api/test/mock_webrtcsession.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timedelta.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/base/timing.h"
 #include "webrtc/media/base/fakemediaengine.h"
 
 using testing::Return;
 using testing::ReturnRef;
 
 namespace webrtc {
 
 namespace {
 
 const int64_t kGetStatsReportTimeoutMs = 1000;
@@ skipping 266 matching lines @@
   EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
   EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);
   EXPECT_TRUE_WAIT(c, kGetStatsReportTimeoutMs);
   EXPECT_EQ(a.get(), b.get());
   // The act of doing |AdvanceTime| processes all messages. If this was not the
   // case we might not require |c| to be fresher than |b|.
   EXPECT_NE(c.get(), b.get());
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) {
-  int64_t before = static_cast<int64_t>(
-      rtc::Timing::WallTimeNow() * rtc::kNumMicrosecsPerSec);
+  int64_t before = rtc::TimeUTCMicros();
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
-  int64_t after = static_cast<int64_t>(
-      rtc::Timing::WallTimeNow() * rtc::kNumMicrosecsPerSec);
+  int64_t after = rtc::TimeUTCMicros();
   EXPECT_EQ(report->GetStatsOfType<RTCPeerConnectionStats>().size(),
             static_cast<size_t>(1)) << "Expecting 1 RTCPeerConnectionStats.";
   const RTCStats* stats = report->Get("RTCPeerConnection");
   EXPECT_TRUE(stats);
   EXPECT_LE(before, stats->timestamp_us());
   EXPECT_LE(stats->timestamp_us(), after);
   {
     // Expected stats with no data channels
     const RTCPeerConnectionStats& pcstats =
         stats->cast_to<RTCPeerConnectionStats>();
@@ skipping 42 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc