Delete all use of tick_util.h.

webrtc/video/vie_sync_module.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/video/vie_sync_module.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
 #include "webrtc/video_frame.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
@@ skipping 23 matching lines @@
 }
 }  // namespace
 
 ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
     : vcm_(vcm),
       clock_(Clock::GetRealTimeClock()),
       video_receiver_(nullptr),
       video_rtp_rtcp_(nullptr),
       voe_channel_id_(-1),
       voe_sync_interface_(nullptr),
-      last_sync_time_(TickTime::Now()),
+      last_sync_time_(rtc::TimeNanos()),
       sync_() {}
 
 ViESyncModule::~ViESyncModule() {
 }
 
 void ViESyncModule::ConfigureSync(int voe_channel_id,
                                   VoEVideoSync* voe_sync_interface,
                                   RtpRtcp* video_rtcp_module,
                                   RtpReceiver* video_receiver) {
   if (voe_channel_id != -1)
     RTC_DCHECK(voe_sync_interface);
   rtc::CritScope lock(&data_cs_);
   // Prevent expensive no-ops.
   if (voe_channel_id_ == voe_channel_id &&
       voe_sync_interface_ == voe_sync_interface &&
       video_receiver_ == video_receiver &&
       video_rtp_rtcp_ == video_rtcp_module) {
     return;
   }
   voe_channel_id_ = voe_channel_id;
   voe_sync_interface_ = voe_sync_interface;
   video_receiver_ = video_receiver;
   video_rtp_rtcp_ = video_rtcp_module;
   sync_.reset(
       new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
 }
 
 int64_t ViESyncModule::TimeUntilNextProcess() {
   const int64_t kSyncIntervalMs = 1000;
-  return kSyncIntervalMs - (TickTime::Now() - last_sync_time_).Milliseconds();
+  return kSyncIntervalMs -
+      (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
 }
 
 void ViESyncModule::Process() {
   rtc::CritScope lock(&data_cs_);
-  last_sync_time_ = TickTime::Now();
+  last_sync_time_ = rtc::TimeNanos();
 
   const int current_video_delay_ms = vcm_->Delay();
 
   if (voe_channel_id_ == -1) {
     return;
   }
   assert(video_rtp_rtcp_ && voe_sync_interface_);
   assert(sync_.get());
 
   int audio_jitter_buffer_delay_ms = 0;
@@ skipping 80 matching lines @@
   int64_t time_to_render_ms =
       frame.render_time_ms() - clock_->TimeInMilliseconds();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   return true;
 }
 
 }  // namespace webrtc