Fix unit for logged bitrates at the end of a call.
BUG=webrtc:5283
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2505873002 .
Cr-Commit-Position: refs/heads/master@{#15100}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 2f69cf4..baa8936 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -369,7 +369,7 @@
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, "
+ LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBytesPerSec, "
<< video_bytes_per_sec.ToString();
}
AggregatedStats audio_bytes_per_sec =
@@ -377,7 +377,7 @@
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, "
+ LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBytesPerSec, "
<< audio_bytes_per_sec.ToString();
}
AggregatedStats rtcp_bytes_per_sec =
@@ -385,7 +385,7 @@
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, "
+ LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBytesPerSec, "
<< rtcp_bytes_per_sec.ToString();
}
AggregatedStats recv_bytes_per_sec =
@@ -393,7 +393,7 @@
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, "
+ LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBytesPerSec, "
<< recv_bytes_per_sec.ToString();
}
}