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Unified Diff: webrtc/call/call.cc

Issue 1530913002: Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years ago
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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);
}
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