Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 9e56b371c819ee92699609ae3e57aadb46e88bea..f8611398c3a5dfe95c4e376b0caa802a911a05a2 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -253,12 +253,12 @@ void Call::UpdateSendHistograms() { |
estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_; |
int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_; |
if (send_bitrate_kbps > 0) { |
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps", |
- send_bitrate_kbps); |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.EstimatedSendBitrateInKbps", |
+ send_bitrate_kbps); |
} |
if (pacer_bitrate_kbps > 0) { |
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps", |
- pacer_bitrate_kbps); |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.PacerBitrateInKbps", |
+ pacer_bitrate_kbps); |
} |
} |
@@ -273,18 +273,18 @@ void Call::UpdateReceiveHistograms() { |
int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000; |
int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec; |
if (video_bitrate_kbps > 0) { |
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps", |
- video_bitrate_kbps); |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.VideoBitrateReceivedInKbps", |
+ video_bitrate_kbps); |
} |
if (audio_bitrate_kbps > 0) { |
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps", |
- audio_bitrate_kbps); |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.AudioBitrateReceivedInKbps", |
+ audio_bitrate_kbps); |
} |
if (rtcp_bitrate_bps > 0) { |
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps", |
- rtcp_bitrate_bps); |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.RtcpBitrateReceivedInBps", |
+ rtcp_bitrate_bps); |
} |
- RTC_HISTOGRAM_COUNTS_100000( |
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000( |
"WebRTC.Call.BitrateReceivedInKbps", |
audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000); |
} |