commit 43cb716e55eab1cf1ae8afd2ac79b51a604d0fa5
author asapersson <asapersson@webrtc.org> Tue Nov 15 08:20:48 2016 -0800
committer Commit bot <commit-bot@chromium.org> Tue Nov 15 16:20:54 2016 +0000
tree e07ce0d67657f6d22937c100b41e0e587650e4e1
parent 841de6a47e6eb1b190ef6a9b1203ecc53c25f6a3

Add ToString method to AggregatedStats and log stats at the end of a call.

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2494423002
Cr-Commit-Position: refs/heads/master@{#15088}

webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3ed7947..2f69cf4 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -349,12 +349,16 @@
   if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
                                 send_bitrate_stats.average);
+    LOG(LS_INFO) << "WebRTC.Call.EstimatedSendBitrateInKbps, "
+                 << send_bitrate_stats.ToString();
   }
   AggregatedStats pacer_bitrate_stats =
       pacer_bitrate_kbps_counter_.ProcessAndGetStats();
   if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
                                 pacer_bitrate_stats.average);
+    LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, "
+                 << pacer_bitrate_stats.ToString();
   }
 }
 
@@ -365,24 +369,32 @@
   if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
                                 video_bytes_per_sec.average * 8 / 1000);
+    LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInKbps, "
+                 << video_bytes_per_sec.ToString();
   }
   AggregatedStats audio_bytes_per_sec =
       received_audio_bytes_per_second_counter_.GetStats();
   if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
                                 audio_bytes_per_sec.average * 8 / 1000);
+    LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInKbps, "
+                 << audio_bytes_per_sec.ToString();
   }
   AggregatedStats rtcp_bytes_per_sec =
       received_rtcp_bytes_per_second_counter_.GetStats();
   if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
                                 rtcp_bytes_per_sec.average * 8);
+    LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
+                 << rtcp_bytes_per_sec.ToString();
   }
   AggregatedStats recv_bytes_per_sec =
       received_bytes_per_second_counter_.GetStats();
   if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
                                 recv_bytes_per_sec.average * 8 / 1000);
+    LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInKbps, "
+                 << recv_bytes_per_sec.ToString();
   }
 }