Delete all use of tick_util.h.

webrtc/common_video/incoming_video_stream.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/common_video/include/incoming_video_stream.h"
 
 #include <assert.h>
 
 #if defined(_WIN32)
 #include <windows.h>
 #elif defined(WEBRTC_LINUX)
 #include <sys/time.h>
 #include <time.h>
 #else
 #include <sys/time.h>
 #endif
 
 #include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/common_video/video_render_frames.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 namespace webrtc {
 
 IncomingVideoStream::IncomingVideoStream(uint32_t stream_id,
                                          bool disable_prerenderer_smoothing)
     : stream_id_(stream_id),
       disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
       incoming_render_thread_(),
       deliver_buffer_event_(EventTimerWrapper::Create()),
@@ skipping 21 matching lines @@
 int32_t IncomingVideoStream::RenderFrame(const uint32_t stream_id,
                                          const VideoFrame& video_frame) {
   rtc::CritScope csS(&stream_critsect_);
 
   if (!running_) {
     return -1;
   }
 
   // Rate statistics.
   num_frames_since_last_calculation_++;
-  int64_t now_ms = TickTime::MillisecondTimestamp();
+  int64_t now_ms = rtc::Time64();
   if (now_ms >= last_rate_calculation_time_ms_ + kFrameRatePeriodMs) {
     incoming_rate_ =
         static_cast<uint32_t>(1000 * num_frames_since_last_calculation_ /
                               (now_ms - last_rate_calculation_time_ms_));
     num_frames_since_last_calculation_ = 0;
     last_rate_calculation_time_ms_ = now_ms;
   }
 
   // Hand over or insert frame.
   if (disable_prerenderer_smoothing_) {
@@ skipping 144 matching lines @@
 void IncomingVideoStream::DeliverFrame(const VideoFrame& video_frame) {
   rtc::CritScope cs(&thread_critsect_);
   if (video_frame.IsZeroSize()) {
     if (render_callback_) {
       if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
         // We have not rendered anything and have a start image.
         temp_frame_.CopyFrame(start_image_);
         render_callback_->RenderFrame(stream_id_, temp_frame_);
       } else if (!timeout_image_.IsZeroSize() &&
                  last_render_time_ms_ + timeout_time_ <
-                     TickTime::MillisecondTimestamp()) {
+                 rtc::Time64()) {
         // Render a timeout image.
         temp_frame_.CopyFrame(timeout_image_);
         render_callback_->RenderFrame(stream_id_, temp_frame_);
       }
     }
 
     // No frame.
     return;
   }
 
   // Send frame for rendering.
   if (external_callback_) {
     external_callback_->RenderFrame(stream_id_, video_frame);
   } else if (render_callback_) {
     render_callback_->RenderFrame(stream_id_, video_frame);
   }
 
   // We're done with this frame.
   last_render_time_ms_ = video_frame.render_time_ms();
 }
 
 }  // namespace webrtc