Add some unit tests to ReceiveStatsticsProxy.

webrtc/video/receive_statistics_proxy.cc

Index: webrtc/video/receive_statistics_proxy.cc
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index c5fa7c004f112400e1b8e56c686cec510232e9d4..26fdae65f9cc2559298e722c45a042e53625783a 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -104,12 +104,14 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
     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(
@@ -121,10 +123,13 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   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) {
@@ -142,30 +147,37 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
         (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)
@@ -175,6 +187,7 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   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 =
@@ -204,7 +217,7 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
   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 /