Add some unit tests to ReceiveStatsticsProxy.

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 86 matching lines @@
       "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 
   if (first_report_block_time_ms_ != -1 &&
       ((clock_->TimeInMilliseconds() - first_report_block_time_ms_) / 1000) >=
           metrics::kMinRunTimeInSeconds) {
     int fraction_lost = report_block_stats_.FractionLostInPercent();
     if (fraction_lost != -1) {
       RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
                                fraction_lost);
+      LOG(LS_INFO) << "WebRTC.Video.ReceivedPacketsLostInPercent "
+                   << fraction_lost;

[sprang_webrtc, 2017/05/11 09:17:57]

Is there any way we could add func to RTC_HISTOGRAM_PERCENTAGE and the other
macros so they do this logging instead? Feels like a pain to have to duplicate
these histogram names everywhere...

[åsapersson, 2017/05/11 09:50:00]

There used to be a RTC_LOGGED_* macro but it has been removed:
https://codereview.webrtc.org/1788783002

[sprang_webrtc, 2017/05/11 10:54:58]

Acknowledged.

     }
   }
 
   const int kMinRequiredSamples = 200;
   int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
-  if (samples > kMinRequiredSamples) {
+  if (samples >= kMinRequiredSamples) {
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
                              round(render_fps_tracker_.ComputeTotalRate()));
     RTC_HISTOGRAM_COUNTS_100000(
         "WebRTC.Video.RenderSqrtPixelsPerSecond",
         round(render_pixel_tracker_.ComputeTotalRate()));
   }
   int width = render_width_counter_.Avg(kMinRequiredSamples);
   int height = render_height_counter_.Avg(kMinRequiredSamples);
   if (width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
+    LOG(LS_INFO) << "WebRTC.Video.ReceivedWidthInPixels " << width;
+    LOG(LS_INFO) << "WebRTC.Video.ReceivedHeightInPixels " << height;
   }
   int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
   if (sync_offset_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
+    LOG(LS_INFO) << "WebRTC.Video.AVSyncOffsetInMs " << sync_offset_ms;
   }
   AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
   if (freq_offset_stats.num_samples > 0) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
                                freq_offset_stats.average);
     LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
                  << freq_offset_stats.ToString();
   }
 
   int num_total_frames =
       stats_.frame_counts.key_frames + stats_.frame_counts.delta_frames;
   if (num_total_frames >= kMinRequiredSamples) {
     int num_key_frames = stats_.frame_counts.key_frames;
     int key_frames_permille =
         (num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
                               key_frames_permille);
+    LOG(LS_INFO) << "WebRTC.Video.KeyFramesReceivedInPermille "
+                 << key_frames_permille;
   }
 
   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
-  if (qp != -1)
+  if (qp != -1) {
     RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
-
+    LOG(LS_INFO) << "WebRTC.Video.Decoded.Vp8.Qp " << qp;
+  }
   int decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
-  if (decode_ms != -1)
+  if (decode_ms != -1) {
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
-
+    LOG(LS_INFO) << "WebRTC.Video.DecodeTimeInMs " << decode_ms;
+  }
   int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredSamples);
   if (jb_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
                                jb_delay_ms);
+    LOG(LS_INFO) << "WebRTC.Video.JitterBufferDelayInMs " << jb_delay_ms;
   }
 
   int target_delay_ms = target_delay_counter_.Avg(kMinRequiredSamples);
   if (target_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
+    LOG(LS_INFO) << "WebRTC.Video.TargetDelayInMs " << target_delay_ms;
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
+    LOG(LS_INFO) << "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_video = e2e_delay_counter_video_.Avg(kMinRequiredSamples);
   if (e2e_delay_ms_video != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs",
                                e2e_delay_ms_video);
+    LOG(LS_INFO) << "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) {
+  if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.BitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                          1000));
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.MediaBitrateReceivedInKbps",
         static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.PaddingBitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
@@ skipping 368 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