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Unified Diff: webrtc/call/call.cc

Issue 1788783002: Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3fd7a931d0cecc88cff0b30a47557db5a21ed586..63ab6e02108a58f7b5f83c2e8c6b559b3674d3be 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -268,12 +268,12 @@ 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_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+ send_bitrate_kbps);
}
if (pacer_bitrate_kbps > 0) {
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
- pacer_bitrate_kbps);
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
+ pacer_bitrate_kbps);
}
}
@@ -288,18 +288,18 @@ 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_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+ video_bitrate_kbps);
}
if (audio_bitrate_kbps > 0) {
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
- audio_bitrate_kbps);
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+ audio_bitrate_kbps);
}
if (rtcp_bitrate_bps > 0) {
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
- rtcp_bitrate_bps);
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+ rtcp_bitrate_bps);
}
- RTC_HISTOGRAM_COUNTS_100000(
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000(
"WebRTC.Call.BitrateReceivedInKbps",
audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
}
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