Delete VideoFrame::set_render_time_ms.

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.
  */
 
@@ skipping 441 matching lines @@
     stats_proxy_->SetResolutionRestrictionStats(
         false, scale_counter_[kCpu] > 0, scale_counter_[kQuality]);
   }
 }
 
 void ViEEncoder::OnFrame(const VideoFrame& video_frame) {
   RTC_DCHECK_RUNS_SERIALIZED(&incoming_frame_race_checker_);
   VideoFrame incoming_frame = video_frame;
 
   // Local time in webrtc time base.
-  int64_t current_time = clock_->TimeInMilliseconds();
-  incoming_frame.set_render_time_ms(current_time);
+  int64_t current_time_us = clock_->TimeInMicroseconds();
+  int64_t current_time_ms = current_time_us / rtc::kNumMicrosecsPerMillisec;
+  // TODO(nisse): This always overrides the incoming timestamp. Don't
+  // do that, trust the frame source.
+  incoming_frame.set_timestamp_us(current_time_us);
 
   // Capture time may come from clock with an offset and drift from clock_.
   int64_t capture_ntp_time_ms;
   if (video_frame.ntp_time_ms() != 0) {
     capture_ntp_time_ms = video_frame.ntp_time_ms();
   } else if (video_frame.render_time_ms() != 0) {
     capture_ntp_time_ms = video_frame.render_time_ms() + delta_ntp_internal_ms_;
   } else {
-    capture_ntp_time_ms = current_time + delta_ntp_internal_ms_;
+    capture_ntp_time_ms = current_time_ms + delta_ntp_internal_ms_;
   }
   incoming_frame.set_ntp_time_ms(capture_ntp_time_ms);
 
   // Convert NTP time, in ms, to RTP timestamp.
   const int kMsToRtpTimestamp = 90;
   incoming_frame.set_timestamp(
       kMsToRtpTimestamp * static_cast<uint32_t>(incoming_frame.ntp_time_ms()));
 
   if (incoming_frame.ntp_time_ms() <= last_captured_timestamp_) {
     // We don't allow the same capture time for two frames, drop this one.
     LOG(LS_WARNING) << "Same/old NTP timestamp ("
                     << incoming_frame.ntp_time_ms()
                     << " <= " << last_captured_timestamp_
                     << ") for incoming frame. Dropping.";
     return;
   }
 
   bool log_stats = false;
-  if (current_time - last_frame_log_ms_ > kFrameLogIntervalMs) {
-    last_frame_log_ms_ = current_time;
+  if (current_time_ms - last_frame_log_ms_ > kFrameLogIntervalMs) {
+    last_frame_log_ms_ = current_time_ms;
     log_stats = true;
   }
 
   last_captured_timestamp_ = incoming_frame.ntp_time_ms();
   encoder_queue_.PostTask(std::unique_ptr<rtc::QueuedTask>(new EncodeTask(
       incoming_frame, this, clock_->TimeInMilliseconds(), log_stats)));
 }
 
 bool ViEEncoder::EncoderPaused() const {
   RTC_DCHECK_RUN_ON(&encoder_queue_);
@@ skipping 251 matching lines @@
   --scale_counter_[reason];
   source_proxy_->RequestHigherResolutionThan(current_pixel_count);
   LOG(LS_INFO) << "Scaling up resolution.";
   for (size_t i = 0; i < kScaleReasonSize; ++i) {
     LOG(LS_INFO) << "Scaled " << scale_counter_[i]
                  << " times for reason: " << (i ? "cpu" : "quality");
   }
 }
 
 }  // namespace webrtc