Add received interframe delay UMA metrics

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 59 matching lines @@
       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),
       total_byte_tracker_(100, 10u),  // bucket_interval_ms, bucket_count
       e2e_delay_max_ms_video_(-1),
       e2e_delay_max_ms_screenshare_(-1),
+      interframe_delay_max_ms_video_(-1),
+      interframe_delay_max_ms_screenshare_(-1),
       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
       first_report_block_time_ms_(-1),
       avg_rtt_ms_(0),
       last_content_type_(VideoContentType::UNSPECIFIED) {
   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 113 matching lines @@
     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);
   }
 
+  int interframe_delay_ms_screenshare =
+      interframe_delay_counter_screenshare_.Avg(kMinRequiredSamples);
+  if (interframe_delay_ms_screenshare != -1) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.InterframeDelayInMs",
+                               interframe_delay_ms_screenshare);
+  }
+
+  int interframe_delay_ms_video =
+      interframe_delay_counter_video_.Avg(kMinRequiredSamples);
+  if (interframe_delay_ms_video != -1) {
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InterframeDelayInMs",
+                               interframe_delay_ms_video);
+  }
+
+  int interframe_delay_max_ms_screenshare =
+      interframe_delay_max_ms_screenshare_;
+  if (interframe_delay_max_ms_screenshare != -1) {
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.Screenshare.InterframeDelayMaxInMs",
+        interframe_delay_ms_screenshare);
+  }
+
+  int interframe_delay_max_ms_video = interframe_delay_max_ms_video_;
+  if (interframe_delay_max_ms_video != -1) {
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.InterframeDelayMaxInMs",
+        interframe_delay_ms_video);
+  }
+
+
   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) {
     RTC_HISTOGRAM_COUNTS_10000(
@@ skipping 286 matching lines @@
       stats_.qp_sum = rtc::Optional<uint64_t>(0);
     }
     *stats_.qp_sum += *qp;
   } else if (stats_.qp_sum) {
     LOG(LS_WARNING)
         << "QP sum was already set and no QP was given for a frame.";
     stats_.qp_sum = rtc::Optional<uint64_t>();
   }
   last_content_type_ = content_type;
   decode_fps_estimator_.Update(1, now);
+  if (last_decoded_frame_time_ms_) {
+    int64_t interframe_delay_ms = now - *last_decoded_frame_time_ms_;
+    RTC_DCHECK_GE(interframe_delay_ms, 0);
+    if (last_content_type_ == VideoContentType::SCREENSHARE) {
+      interframe_delay_counter_screenshare_.Add(interframe_delay_ms);
+      if (interframe_delay_max_ms_screenshare_ < interframe_delay_ms) {
+        interframe_delay_max_ms_screenshare_ = interframe_delay_ms;
+      }
+    } else {
+      interframe_delay_counter_video_.Add(interframe_delay_ms);
+      if (interframe_delay_max_ms_video_ < interframe_delay_ms) {
+        interframe_delay_max_ms_video_ = interframe_delay_ms;
+      }
+    }
+  }
+  last_decoded_frame_time_ms_.emplace(now);
 }
 
 void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
   int width = frame.width();
   int height = frame.height();
   RTC_DCHECK_GT(width, 0);
   RTC_DCHECK_GT(height, 0);
   uint64_t now = clock_->TimeInMilliseconds();
 
   rtc::CritScope lock(&crit_);
@@ skipping 101 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