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

webrtc/stats/rtcstatscollector.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 <utility>
 #include <vector>
 
 #include "webrtc/api/peerconnection.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/timing.h"
 
 namespace webrtc {
 
 rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
     PeerConnection* pc, int64_t cache_lifetime_us) {
   return rtc::scoped_refptr<RTCStatsCollector>(
       new rtc::RefCountedObject<RTCStatsCollector>(pc, cache_lifetime_us));
 }
 
 RTCStatsCollector::RTCStatsCollector(PeerConnection* pc,
@@ skipping 26 matching lines @@
     // We have a fresh cached report to deliver.
     DeliverCachedReport();
   } else if (!num_pending_partial_reports_) {
     // Only start gathering stats if we're not already gathering stats. In the
     // case of already gathering stats, |callback_| will be invoked when there
     // are no more pending partial reports.
 
     // "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970,
     // UTC), in microseconds. The system clock could be modified and is not
     // necessarily monotonically increasing.
-    int64_t timestamp_us = static_cast<int64_t>(
-        rtc::Timing::WallTimeNow() * rtc::kNumMicrosecsPerSec);
+    int64_t timestamp_us = rtc::TimeUTCMicros();
 
     num_pending_partial_reports_ = 3;
     partial_report_timestamp_us_ = cache_now_us;
     invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread_,
         rtc::Bind(&RTCStatsCollector::ProducePartialResultsOnSignalingThread,
             rtc::scoped_refptr<RTCStatsCollector>(this), timestamp_us));
     invoker_.AsyncInvoke<void>(RTC_FROM_HERE, worker_thread_,
         rtc::Bind(&RTCStatsCollector::ProducePartialResultsOnWorkerThread,
             rtc::scoped_refptr<RTCStatsCollector>(this), timestamp_us));
     invoker_.AsyncInvoke<void>(RTC_FROM_HERE, network_thread_,
@@ skipping 94 matching lines @@
   // constant.
   std::unique_ptr<RTCPeerConnectionStats> stats(
     new RTCPeerConnectionStats("RTCPeerConnection", timestamp_us));
   stats->data_channels_opened = data_channels_opened;
   stats->data_channels_closed = static_cast<uint32_t>(data_channels.size()) -
                                 data_channels_opened;
   return stats;
 }
 
 }  // namespace webrtc