Delete all use of tick_util.h.

webrtc/video/vie_encoder.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_encoder.h"
 
 #include <assert.h>
 
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/video/overuse_frame_detector.h"
 #include "webrtc/video/send_statistics_proxy.h"
 #include "webrtc/video_frame.h"
 
 namespace webrtc {
 
 static const float kStopPaddingThresholdMs = 2000;
 
 class QMVideoSettingsCallback : public VCMQMSettingsCallback {
  public:
@@ skipping 166 matching lines @@
     }
   }
 
   // Disable padding if only sending one stream and video isn't suspended and
   // min-transmit bitrate isn't used (applied later).
   if (!video_is_suspended && send_codec.numberOfSimulcastStreams <= 1)
     pad_up_to_bitrate_bps = 0;
 
   // The amount of padding should decay to zero if no frames are being
   // captured/encoded unless a min-transmit bitrate is used.
-  int64_t now_ms = TickTime::MillisecondTimestamp();
+  int64_t now_ms = rtc::TimeMillis();
   if (now_ms - time_of_last_frame_activity_ms > kStopPaddingThresholdMs)
     pad_up_to_bitrate_bps = 0;
 
   // Pad up to min bitrate.
   if (pad_up_to_bitrate_bps < min_transmit_bitrate_bps)
     pad_up_to_bitrate_bps = min_transmit_bitrate_bps;
 
   // Padding may never exceed bitrate estimate.
   if (pad_up_to_bitrate_bps > bitrate_bps)
     pad_up_to_bitrate_bps = bitrate_bps;
@@ skipping 28 matching lines @@
   if (encoder_paused_and_dropped_frame_) {
     TRACE_EVENT_ASYNC_END0("webrtc", "EncoderPaused", this);
   }
   encoder_paused_and_dropped_frame_ = false;
 }
 
 void ViEEncoder::EncodeVideoFrame(const VideoFrame& video_frame) {
   VideoCodecType codec_type;
   {
     rtc::CritScope lock(&data_cs_);
-    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+    time_of_last_frame_activity_ms_ = rtc::TimeMillis();
     if (EncoderPaused()) {
       TraceFrameDropStart();
       return;
     }
     TraceFrameDropEnd();
     codec_type = encoder_config_.codecType;
   }
 
   TRACE_EVENT_ASYNC_STEP0("webrtc", "Video", video_frame.render_time_ms(),
                           "Encode");
@@ skipping 55 matching lines @@
 void ViEEncoder::OnSetRates(uint32_t bitrate_bps, int framerate) {
   if (stats_proxy_)
     stats_proxy_->OnSetRates(bitrate_bps, framerate);
 }
 
 int32_t ViEEncoder::Encoded(const EncodedImage& encoded_image,
                             const CodecSpecificInfo* codec_specific_info,
                             const RTPFragmentationHeader* fragmentation) {
   {
     rtc::CritScope lock(&data_cs_);
-    time_of_last_frame_activity_ms_ = TickTime::MillisecondTimestamp();
+    time_of_last_frame_activity_ms_ = rtc::TimeMillis();
   }
   if (stats_proxy_) {
     stats_proxy_->OnSendEncodedImage(encoded_image, codec_specific_info);
   }
 
   int success = 0;
   {
     rtc::CritScope lock(&sink_cs_);
     success = sink_->Encoded(encoded_image, codec_specific_info, fragmentation);
   }
@@ skipping 61 matching lines @@
 }
 
 int32_t QMVideoSettingsCallback::SetVideoQMSettings(
     const uint32_t frame_rate,
     const uint32_t width,
     const uint32_t height) {
   return vp_->SetTargetResolution(width, height, frame_rate);
 }
 
 }  // namespace webrtc