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);

[nisse-webrtc, 2017/01/13 09:33:24]

I plan to delete this in a followup. This line (both before and after this cl)
means that we discard the timestamp attached earlier in the pipeline, which is
silly. But fixing it may break video sources which doesn't get it right, so
might need some extra care.

[perkj_webrtc, 2017/01/13 10:39:28]

doesn't that cause a regression with for example the Logitech camera? 

 
   // 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