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

webrtc/api/statscollector.cc

 /*
  *  Copyright 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/api/statscollector.h"
 
 #include <memory>
 #include <utility>
 #include <vector>
 
+// StatsCollector::GetTimeNow
+#if defined(WEBRTC_POSIX)
+#include <sys/time.h>
+#elif defined(WEBRTC_WIN)
+#include <sys/timeb.h>
+#endif
+
 #include "webrtc/api/peerconnection.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/timing.h"
 #include "webrtc/pc/channel.h"
 
 namespace webrtc {
 namespace {
 
 // The following is the enum RTCStatsIceCandidateType from
 // http://w3c.github.io/webrtc-stats/#rtcstatsicecandidatetype-enum such that
 // our stats report for ice candidate type could conform to that.
 const char STATSREPORT_LOCAL_PORT_TYPE[] = "host";
 const char STATSREPORT_STUN_PORT_TYPE[] = "serverreflexive";
@@ skipping 338 matching lines @@
 StatsCollector::StatsCollector(PeerConnection* pc)
     : pc_(pc), stats_gathering_started_(0) {
   RTC_DCHECK(pc_);
 }
 
 StatsCollector::~StatsCollector() {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
 }
 
 double StatsCollector::GetTimeNow() {
-  return rtc::Timing::WallTimeNow() * rtc::kNumMillisecsPerSec;
+  // TODO(nisse): Helper function belongs in timeutils.h.

[hbos, 2016/08/30 12:56:40]

I think this TODO should be fixed in this CL before landing it.

[nisse-webrtc, 2016/08/30 13:20:25]

Agreed. We only have to settle on a name.

+#if defined(WEBRTC_POSIX)
+  struct timeval time;
+  gettimeofday(&time, NULL);
+  // Convert from second (1.0) and microsecond (1e-6).
+  return (static_cast<double>(time.tv_sec) +
+          static_cast<double>(time.tv_usec) * 1.0e-6);
+
+#elif defined(WEBRTC_WIN)
+  struct _timeb time;
+  _ftime(&time);
+  // Convert from second (1.0) and milliseconds (1e-3).
+  return (static_cast<double>(time.time) +
+          static_cast<double>(time.millitm) * 1.0e-3);
+#endif
 }
 
 // Adds a MediaStream with tracks that can be used as a |selector| in a call
 // to GetStats.
 void StatsCollector::AddStream(MediaStreamInterface* stream) {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
   RTC_DCHECK(stream != NULL);
 
   CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(),
                                        &reports_, track_ids_);
@@ skipping 571 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc