| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 84575723958fc7217efcf1850fee2672a4fc21d1..2cc296dd36e9de80417f817d38e66afae6ccee18 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -312,7 +312,7 @@ Call::~Call() {
|
| }
|
|
|
| void Call::UpdateHistograms() {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000(
|
| + RTC_HISTOGRAM_COUNTS_100000(
|
| "WebRTC.Call.LifetimeInSeconds",
|
| (clock_->TimeInMilliseconds() - start_ms_) / 1000);
|
| }
|
| @@ -328,14 +328,14 @@ void Call::UpdateSendHistograms() {
|
| AggregatedStats send_bitrate_stats =
|
| estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
|
| if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
|
| - send_bitrate_stats.average);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
|
| + send_bitrate_stats.average);
|
| }
|
| AggregatedStats pacer_bitrate_stats =
|
| pacer_bitrate_kbps_counter_.ProcessAndGetStats();
|
| if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
|
| - pacer_bitrate_stats.average);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
|
| + pacer_bitrate_stats.average);
|
| }
|
| }
|
|
|
| @@ -344,26 +344,26 @@ void Call::UpdateReceiveHistograms() {
|
| AggregatedStats video_bytes_per_sec =
|
| received_video_bytes_per_second_counter_.GetStats();
|
| if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
|
| - video_bytes_per_sec.average * 8 / 1000);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
|
| + video_bytes_per_sec.average * 8 / 1000);
|
| }
|
| AggregatedStats audio_bytes_per_sec =
|
| received_audio_bytes_per_second_counter_.GetStats();
|
| if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
|
| - audio_bytes_per_sec.average * 8 / 1000);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
|
| + audio_bytes_per_sec.average * 8 / 1000);
|
| }
|
| AggregatedStats rtcp_bytes_per_sec =
|
| received_rtcp_bytes_per_second_counter_.GetStats();
|
| if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
|
| - rtcp_bytes_per_sec.average * 8);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
|
| + rtcp_bytes_per_sec.average * 8);
|
| }
|
| AggregatedStats recv_bytes_per_sec =
|
| received_bytes_per_second_counter_.GetStats();
|
| if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
|
| - RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
|
| - recv_bytes_per_sec.average * 8 / 1000);
|
| + RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
|
| + recv_bytes_per_sec.average * 8 / 1000);
|
| }
|
| }
|
|
|
|
|