Add content type information to encoded images and corresponding rtp extension header

webrtc/video/receive_statistics_proxy.cc

 /*
  *  Copyright (c) 2013 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 55 matching lines @@
                           kHighVarianceThreshold,
                           kBadFraction,
                           kNumMeasurementsVariance),
       num_bad_states_(0),
       num_certain_states_(0),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
       render_fps_tracker_(100, 10u),
       render_pixel_tracker_(100, 10u),
+      e2e_delay_max_ms_video_(-1),
+      e2e_delay_max_ms_screenshare_(-1),

[sprang_webrtc, 2017/03/28 11:14:23]

Maybe we should report all stats and reset state if we detect a content type
switch, similar to how we do it for the send side?

[ilnik, 2017/03/28 14:47:02]

Problem here is that due to packet loss/reordering we will get some mixed
statistics in the vicinity of switch. We might make this change in the future if
we need some more metrics, it should be a separate CL.

       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
       first_report_block_time_ms_(-1),
       avg_rtt_ms_(0),
       frame_window_accumulated_bytes_(0) {
   stats_.ssrc = config_.rtp.remote_ssrc;
   // TODO(brandtr): Replace |rtx_stats_| with a single instance of
   // StreamDataCounters.
   if (config_.rtp.rtx_ssrc) {
     rtx_stats_[config_.rtp.rtx_ssrc] = StreamDataCounters();
   }
@@ skipping 76 matching lines @@
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
   }
   int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
 
-  int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
-  if (e2e_delay_ms != -1)
-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);
+  int e2e_delay_ms_video = e2e_delay_counter_video_.Avg(kMinRequiredSamples);
+  if (e2e_delay_ms_video != -1) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs",
+                               e2e_delay_ms_video);
+  }
+
+  int e2e_delay_ms_screenshare =
+      e2e_delay_counter_screenshare_.Avg(kMinRequiredSamples);
+  if (e2e_delay_ms_screenshare != -1) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.EndToEndDelayInMs",
+                               e2e_delay_ms_screenshare);
+  }
+
+  int e2e_delay_max_ms_video = e2e_delay_max_ms_video_;
+  if (e2e_delay_max_ms_video != -1) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.EndToEndDelayMaxInMs",
+                                e2e_delay_max_ms_video);
+  }
+
+  int e2e_delay_max_ms_screenshare = e2e_delay_max_ms_screenshare_;
+  if (e2e_delay_max_ms_screenshare != -1) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs",
+                                e2e_delay_max_ms_screenshare);
+  }
 
   StreamDataCounters rtp = stats_.rtp_stats;
   StreamDataCounters rtx;
   for (auto it : rtx_stats_)
     rtx.Add(it.second);
   StreamDataCounters rtp_rtx = rtp;
   rtp_rtx.Add(rtx);
   int64_t elapsed_sec =
       rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
   if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
@@ skipping 277 matching lines @@
   ++stats_.frames_rendered;
   stats_.width = width;
   stats_.height = height;
   render_width_counter_.Add(width);
   render_height_counter_.Add(height);
   render_fps_tracker_.AddSamples(1);
   render_pixel_tracker_.AddSamples(sqrt(width * height));
 
   if (frame.ntp_time_ms() > 0) {
     int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
-    if (delay_ms >= 0)
-      e2e_delay_counter_.Add(delay_ms);
+    if (delay_ms >= 0) {
+      if (frame.content_type() == kVideoContent_Screenshare) {
+        e2e_delay_max_ms_screenshare_ =
+            std::max(delay_ms, e2e_delay_max_ms_screenshare_);
+        e2e_delay_counter_screenshare_.Add(delay_ms);
+      } else {
+        e2e_delay_max_ms_video_ = std::max(delay_ms, e2e_delay_max_ms_video_);
+        e2e_delay_counter_video_.Add(delay_ms);
+      }
+    }
   }
 }
 
 void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms,
                                                  double estimated_freq_khz) {
   rtc::CritScope lock(&crit_);
   sync_offset_counter_.Add(std::abs(sync_offset_ms));
   stats_.sync_offset_ms = sync_offset_ms;
 
   const double kMaxFreqKhz = 10000.0;
@@ skipping 64 matching lines @@
   sum = 0;
 }
 
 void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
                                          int64_t max_rtt_ms) {
   rtc::CritScope lock(&crit_);
   avg_rtt_ms_ = avg_rtt_ms;
 }
 
 }  // namespace webrtc