commit 076c0118c52c0ff74f81dc855b0753e19e5a5010
author asapersson <asapersson@webrtc.org> Wed Nov 30 13:17:16 2016
committer Commit bot <commit-bot@chromium.org> Wed Nov 30 13:17:21 2016
tree 6cd92660513feb1684e069fc876b61cda58b3db7
parent aff96361d3936e34a270f93ca11d0cf868239517

Change unit of logged bitrate stats in bytes/s to bits/s.

Multiplier added to ToString method in AggregatedStats.

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2535323003
Cr-Commit-Position: refs/heads/master@{#15330}

webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ae04409..e76df36 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -383,32 +383,32 @@
   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.VideoBitrateReceivedInBytesPerSec, "
-                 << video_bytes_per_sec.ToString();
+    LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
+                 << video_bytes_per_sec.ToStringWithMultiplier(8);
   }
   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.AudioBitrateReceivedInBytesPerSec, "
-                 << audio_bytes_per_sec.ToString();
+    LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, "
+                 << audio_bytes_per_sec.ToStringWithMultiplier(8);
   }
   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.RtcpBitrateReceivedInBytesPerSec, "
-                 << rtcp_bytes_per_sec.ToString();
+    LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
+                 << rtcp_bytes_per_sec.ToStringWithMultiplier(8);
   }
   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.BitrateReceivedInBytesPerSec, "
-                 << recv_bytes_per_sec.ToString();
+    LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
+                 << recv_bytes_per_sec.ToStringWithMultiplier(8);
   }
 }
 

webrtc/video/stats_counter.cc

diff --git a/webrtc/video/stats_counter.cc b/webrtc/video/stats_counter.cc
index b1c24ee..86dedd9 100644
--- a/webrtc/video/stats_counter.cc
+++ b/webrtc/video/stats_counter.cc
@@ -26,11 +26,15 @@
 }  // namespace
 
 std::string AggregatedStats::ToString() const {
+  return ToStringWithMultiplier(1);
+}
+
+std::string AggregatedStats::ToStringWithMultiplier(int multiplier) const {
   std::stringstream ss;
   ss << "periodic_samples:" << num_samples << ", {";
-  ss << "min:" << min << ", ";
-  ss << "avg:" << average << ", ";
-  ss << "max:" << max << "}";
+  ss << "min:" << (min * multiplier) << ", ";
+  ss << "avg:" << (average * multiplier) << ", ";
+  ss << "max:" << (max * multiplier) << "}";
   return ss.str();
 }
 

webrtc/video/stats_counter.h

diff --git a/webrtc/video/stats_counter.h b/webrtc/video/stats_counter.h
index e84d73b..6dc94b4 100644
--- a/webrtc/video/stats_counter.h
+++ b/webrtc/video/stats_counter.h
@@ -33,6 +33,7 @@
 
 struct AggregatedStats {
   std::string ToString() const;
+  std::string ToStringWithMultiplier(int multiplier) const;
 
   int64_t num_samples = 0;
   int min = -1;