Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 9e56b37..f861139 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -253,12 +253,12 @@
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 @@
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);
}
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
index 0572b26..b434da2 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -97,7 +97,7 @@
if (value != last_value_ || first_time_) {
first_time_ = false;
last_value_ = value;
- RTC_HISTOGRAM_COUNTS_100(histogram_name_, value);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
}
}
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index e6a6fbf..8f87376 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -50,7 +50,7 @@
}
void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
- RTC_HISTOGRAM_COUNTS(uma_name_, value, 1, max_value_, 50);
+ RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
}
StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
@@ -187,9 +187,9 @@
}
void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
- outage_duration_ms, 1 /* min */, 2000 /* max */,
- 100 /* bucket count */);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.DelayedPacketOutageEventMs",
+ outage_duration_ms, 1 /* min */, 2000 /* max */,
+ 100 /* bucket count */);
delayed_packet_outage_counter_.RegisterSample();
}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index a332945..c0c5e8a 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1346,8 +1346,9 @@
capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
if (diff_stream_delay_ms > kMinDiffDelayMs &&
capture_.last_stream_delay_ms != 0) {
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
- diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
+ RTC_HISTOGRAM_COUNTS_SPARSE(
+ "WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
+ kMinDiffDelayMs, 1000, 100);
if (capture_.stream_delay_jumps == -1) {
capture_.stream_delay_jumps = 0; // Activate counter if needed.
}
@@ -1364,9 +1365,9 @@
aec_system_delay_ms - capture_.last_aec_system_delay_ms;
if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
capture_.last_aec_system_delay_ms != 0) {
- RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
- diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
- 100);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
+ diff_aec_system_delay_ms, kMinDiffDelayMs,
+ 1000, 100);
if (capture_.aec_system_delay_jumps == -1) {
capture_.aec_system_delay_jumps = 0; // Activate counter if needed.
}
@@ -1382,7 +1383,7 @@
rtc::CritScope cs_capture(&crit_capture_);
if (capture_.stream_delay_jumps > -1) {
- RTC_HISTOGRAM_ENUMERATION(
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
capture_.stream_delay_jumps, 51);
}
@@ -1390,8 +1391,8 @@
capture_.last_stream_delay_ms = 0;
if (capture_.aec_system_delay_jumps > -1) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
- capture_.aec_system_delay_jumps, 51);
+ RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
+ capture_.aec_system_delay_jumps, 51);
}
capture_.aec_system_delay_jumps = -1;
capture_.last_aec_system_delay_ms = 0;
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 96a3b47..258c4d9 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -146,8 +146,8 @@
for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
if (!rampup_uma_stats_updated_[i] &&
bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
- RTC_HISTOGRAM_COUNTS_100000(kUmaRampupMetrics[i].metric_name,
- now_ms - first_report_time_ms_);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(kUmaRampupMetrics[i].metric_name,
+ now_ms - first_report_time_ms_);
rampup_uma_stats_updated_[i] = true;
}
}
@@ -156,22 +156,19 @@
} else if (uma_update_state_ == kNoUpdate) {
uma_update_state_ = kFirstDone;
bitrate_at_2_seconds_kbps_ = bitrate_kbps;
- RTC_HISTOGRAM_COUNTS(
- "WebRTC.BWE.InitiallyLostPackets", initially_lost_packets_, 0, 100, 50);
- RTC_HISTOGRAM_COUNTS(
- "WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0, 2000, 50);
- RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
- bitrate_at_2_seconds_kbps_,
- 0,
- 2000,
- 50);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitiallyLostPackets",
+ initially_lost_packets_, 0, 100, 50);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
+ 0, 2000, 50);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialBandwidthEstimate",
+ bitrate_at_2_seconds_kbps_, 0, 2000, 50);
} else if (uma_update_state_ == kFirstDone &&
now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
uma_update_state_ = kDone;
int bitrate_diff_kbps =
std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
- RTC_HISTOGRAM_COUNTS(
- "WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps, 0, 2000, 50);
+ RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialVsConvergedDiff",
+ bitrate_diff_kbps, 0, 2000, 50);
}
}
diff --git a/webrtc/modules/video_coding/jitter_buffer.cc b/webrtc/modules/video_coding/jitter_buffer.cc
index a1142bb..a381880 100644
--- a/webrtc/modules/video_coding/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/jitter_buffer.cc
@@ -281,17 +281,18 @@
return;
}
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
- num_discarded_packets_ * 100 / num_packets_);
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
- num_duplicated_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DiscardedPacketsInPercent",
+ num_discarded_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DuplicatedPacketsInPercent",
+ num_duplicated_packets_ * 100 / num_packets_);
int total_frames =
receive_statistics_.key_frames + receive_statistics_.delta_frames;
if (total_frames > 0) {
- RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.CompleteFramesReceivedPerSecond",
+ RTC_HISTOGRAM_COUNTS_SPARSE_100(
+ "WebRTC.Video.CompleteFramesReceivedPerSecond",
static_cast<int>((total_frames / elapsed_sec) + 0.5f));
- RTC_HISTOGRAM_COUNTS_1000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_1000(
"WebRTC.Video.KeyFramesReceivedInPermille",
static_cast<int>(
(receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
diff --git a/webrtc/modules/video_coding/timing.cc b/webrtc/modules/video_coding/timing.cc
index f1a127a..d2563a4 100644
--- a/webrtc/modules/video_coding/timing.cc
+++ b/webrtc/modules/video_coding/timing.cc
@@ -62,14 +62,16 @@
if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
return;
}
- RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
+ RTC_HISTOGRAM_COUNTS_SPARSE_100(
+ "WebRTC.Video.DecodedFramesPerSecond",
static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+ "WebRTC.Video.DelayedFramesToRenderer",
num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
if (num_delayed_decoded_frames_ > 0) {
- RTC_HISTOGRAM_COUNTS_1000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_1000(
"WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
- sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+ sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
}
}
diff --git a/webrtc/system_wrappers/include/metrics.h b/webrtc/system_wrappers/include/metrics.h
index 2e6e7b7..5e8ca11 100644
--- a/webrtc/system_wrappers/include/metrics.h
+++ b/webrtc/system_wrappers/include/metrics.h
@@ -69,44 +69,43 @@
// Also consider changing string to const char* when switching to atomics.
// Histogram for counters.
-#define RTC_HISTOGRAM_COUNTS_100(name, sample) RTC_HISTOGRAM_COUNTS( \
- name, sample, 1, 100, 50)
+#define RTC_HISTOGRAM_COUNTS_SPARSE_100(name, sample) \
+ RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 100, 50)
-#define RTC_HISTOGRAM_COUNTS_200(name, sample) RTC_HISTOGRAM_COUNTS( \
- name, sample, 1, 200, 50)
+#define RTC_HISTOGRAM_COUNTS_SPARSE_200(name, sample) \
+ RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 200, 50)
-#define RTC_HISTOGRAM_COUNTS_1000(name, sample) RTC_HISTOGRAM_COUNTS( \
- name, sample, 1, 1000, 50)
+#define RTC_HISTOGRAM_COUNTS_SPARSE_1000(name, sample) \
+ RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 1000, 50)
-#define RTC_HISTOGRAM_COUNTS_10000(name, sample) RTC_HISTOGRAM_COUNTS( \
- name, sample, 1, 10000, 50)
+#define RTC_HISTOGRAM_COUNTS_SPARSE_10000(name, sample) \
+ RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 10000, 50)
-#define RTC_HISTOGRAM_COUNTS_100000(name, sample) RTC_HISTOGRAM_COUNTS( \
- name, sample, 1, 100000, 50)
+#define RTC_HISTOGRAM_COUNTS_SPARSE_100000(name, sample) \
+ RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, 1, 100000, 50)
-#define RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count) \
- RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
- webrtc::metrics::HistogramFactoryGetCounts( \
- name, min, max, bucket_count))
+#define RTC_HISTOGRAM_COUNTS_SPARSE(name, sample, min, max, bucket_count) \
+ RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, \
+ webrtc::metrics::HistogramFactoryGetCounts( \
+ name, min, max, bucket_count))
// Histogram for percentage.
-#define RTC_HISTOGRAM_PERCENTAGE(name, sample) \
- RTC_HISTOGRAM_ENUMERATION(name, sample, 101)
+#define RTC_HISTOGRAM_PERCENTAGE_SPARSE(name, sample) \
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, 101)
// Histogram for enumerators.
// |boundary| should be above the max enumerator sample.
-#define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \
- RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
- webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
+#define RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, boundary) \
+ RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, \
+ webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
-#define RTC_HISTOGRAM_COMMON_BLOCK(constant_name, sample, \
- factory_get_invocation) \
+#define RTC_HISTOGRAM_COMMON_BLOCK_SLOW(constant_name, sample, \
+ factory_get_invocation) \
do { \
webrtc::metrics::Histogram* histogram_pointer = factory_get_invocation; \
webrtc::metrics::HistogramAdd(histogram_pointer, constant_name, sample); \
} while (0)
-
namespace webrtc {
namespace metrics {
diff --git a/webrtc/video/receive_statistics_proxy.cc b/webrtc/video/receive_statistics_proxy.cc
index c13c807..d6ab4ff 100644
--- a/webrtc/video/receive_statistics_proxy.cc
+++ b/webrtc/video/receive_statistics_proxy.cc
@@ -37,37 +37,39 @@
void ReceiveStatisticsProxy::UpdateHistograms() {
int fraction_lost = report_block_stats_.FractionLostInPercent();
if (fraction_lost != -1) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
- fraction_lost);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.ReceivedPacketsLostInPercent",
+ fraction_lost);
}
const int kMinRequiredSamples = 200;
int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
if (samples > kMinRequiredSamples) {
- RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
+ RTC_HISTOGRAM_COUNTS_SPARSE_100("WebRTC.Video.RenderFramesPerSecond",
round(render_fps_tracker_.ComputeTotalRate()));
- RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.RenderSqrtPixelsPerSecond",
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Video.RenderSqrtPixelsPerSecond",
round(render_pixel_tracker_.ComputeTotalRate()));
}
int width = render_width_counter_.Avg(kMinRequiredSamples);
int height = render_height_counter_.Avg(kMinRequiredSamples);
if (width != -1) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedWidthInPixels",
+ width);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.ReceivedHeightInPixels",
+ height);
}
int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
if (qp != -1)
- RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
+ RTC_HISTOGRAM_COUNTS_SPARSE_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
// TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
// not per frame. Change decode time to include every frame.
const int kMinRequiredDecodeSamples = 5;
int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
if (decode_ms != -1)
- RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
+ RTC_HISTOGRAM_COUNTS_SPARSE_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
if (delay_ms != -1)
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
diff --git a/webrtc/video/send_statistics_proxy.cc b/webrtc/video/send_statistics_proxy.cc
index c198ad2..0a84439 100644
--- a/webrtc/video/send_statistics_proxy.cc
+++ b/webrtc/video/send_statistics_proxy.cc
@@ -57,7 +57,7 @@
}
void UpdateCodecTypeHistogram(const std::string& payload_name) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.Encoder.CodecType",
+ RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Video.Encoder.CodecType",
PayloadNameToHistogramCodecType(payload_name), kVideoMax);
}
} // namespace
@@ -98,56 +98,63 @@
int in_height = input_height_counter_.Avg(kMinRequiredSamples);
int in_fps = round(input_frame_rate_tracker_.ComputeTotalRate());
if (in_width != -1) {
- RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "InputWidthInPixels", in_width);
- RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "InputHeightInPixels", in_height);
- RTC_HISTOGRAM_COUNTS_100(uma_prefix_ + "InputFramesPerSecond", in_fps);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "InputWidthInPixels",
+ in_width);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "InputHeightInPixels",
+ in_height);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix_ + "InputFramesPerSecond",
+ in_fps);
}
int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
int sent_fps = round(sent_frame_rate_tracker_.ComputeTotalRate());
if (sent_width != -1) {
- RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "SentWidthInPixels", sent_width);
- RTC_HISTOGRAM_COUNTS_10000(uma_prefix_ + "SentHeightInPixels", sent_height);
- RTC_HISTOGRAM_COUNTS_100(uma_prefix_ + "SentFramesPerSecond", sent_fps);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "SentWidthInPixels",
+ sent_width);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(uma_prefix_ + "SentHeightInPixels",
+ sent_height);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix_ + "SentFramesPerSecond",
+ sent_fps);
}
int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
if (encode_ms != -1)
- RTC_HISTOGRAM_COUNTS_1000(uma_prefix_ + "EncodeTimeInMs", encode_ms);
+ RTC_HISTOGRAM_COUNTS_SPARSE_1000(uma_prefix_ + "EncodeTimeInMs", encode_ms);
int key_frames_permille = key_frame_counter_.Permille(kMinRequiredSamples);
if (key_frames_permille != -1) {
- RTC_HISTOGRAM_COUNTS_1000(uma_prefix_ + "KeyFramesSentInPermille",
- key_frames_permille);
+ RTC_HISTOGRAM_COUNTS_SPARSE_1000(uma_prefix_ + "KeyFramesSentInPermille",
+ key_frames_permille);
}
int quality_limited =
quality_limited_frame_counter_.Percent(kMinRequiredSamples);
if (quality_limited != -1) {
- RTC_HISTOGRAM_PERCENTAGE(uma_prefix_ + "QualityLimitedResolutionInPercent",
- quality_limited);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+ uma_prefix_ + "QualityLimitedResolutionInPercent", quality_limited);
}
int downscales = quality_downscales_counter_.Avg(kMinRequiredSamples);
if (downscales != -1) {
- RTC_HISTOGRAM_ENUMERATION(
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
uma_prefix_ + "QualityLimitedResolutionDownscales", downscales, 20);
}
int bw_limited = bw_limited_frame_counter_.Percent(kMinRequiredSamples);
if (bw_limited != -1) {
- RTC_HISTOGRAM_PERCENTAGE(
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
uma_prefix_ + "BandwidthLimitedResolutionInPercent", bw_limited);
}
int num_disabled = bw_resolutions_disabled_counter_.Avg(kMinRequiredSamples);
if (num_disabled != -1) {
- RTC_HISTOGRAM_ENUMERATION(
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
uma_prefix_ + "BandwidthLimitedResolutionsDisabled", num_disabled, 10);
}
int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
if (delay_ms != -1)
- RTC_HISTOGRAM_COUNTS_100000(uma_prefix_ + "SendSideDelayInMs", delay_ms);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix_ + "SendSideDelayInMs",
+ delay_ms);
int max_delay_ms = max_delay_counter_.Avg(kMinRequiredSamples);
if (max_delay_ms != -1) {
- RTC_HISTOGRAM_COUNTS_100000(uma_prefix_ + "SendSideDelayMaxInMs",
- max_delay_ms);
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix_ + "SendSideDelayMaxInMs",
+ max_delay_ms);
}
}
diff --git a/webrtc/video/vie_channel.cc b/webrtc/video/vie_channel.cc
index e39ffbf..401cba8 100644
--- a/webrtc/video/vie_channel.cc
+++ b/webrtc/video/vie_channel.cc
@@ -202,7 +202,7 @@
if (time_of_first_rtt_ms_ != -1 && num_rtts_ > 0 &&
elapsed_sec > metrics::kMinRunTimeInSeconds) {
int64_t avg_rtt_ms = (rtt_sum_ms_ + num_rtts_ / 2) / num_rtts_;
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
}
}
@@ -212,21 +212,24 @@
GetSendRtcpPacketTypeCounter(&rtcp_counter);
int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
- rtcp_counter.nack_packets * 60 / elapsed_sec);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
- rtcp_counter.fir_packets * 60 / elapsed_sec);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
- rtcp_counter.pli_packets * 60 / elapsed_sec);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.NackPacketsReceivedPerMinute",
+ rtcp_counter.nack_packets * 60 / elapsed_sec);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.FirPacketsReceivedPerMinute",
+ rtcp_counter.fir_packets * 60 / elapsed_sec);
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.PliPacketsReceivedPerMinute",
+ rtcp_counter.pli_packets * 60 / elapsed_sec);
if (rtcp_counter.nack_requests > 0) {
- RTC_HISTOGRAM_PERCENTAGE(
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent",
rtcp_counter.UniqueNackRequestsInPercent());
}
int fraction_lost = report_block_stats_sender_->FractionLostInPercent();
if (fraction_lost != -1) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent",
- fraction_lost);
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.SentPacketsLostInPercent",
+ fraction_lost);
}
}
@@ -239,23 +242,23 @@
Clock::GetRealTimeClock()->TimeInMilliseconds()) /
1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
- RTC_HISTOGRAM_COUNTS_100000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(
"WebRTC.Video.BitrateSentInKbps",
static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.MediaBitrateSentInKbps",
static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.PaddingBitrateSentInKbps",
static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.RetransmittedBitrateSentInKbps",
static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
elapsed_sec / 1000));
if (rtp_rtcp_modules_[0]->RtxSendStatus() != kRtxOff) {
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.RtxBitrateSentInKbps",
static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
1000));
@@ -266,9 +269,10 @@
rtp_rtcp_modules_[0]->GenericFECStatus(&fec_enabled, &pltype_red,
&pltype_fec);
if (fec_enabled) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps",
- static_cast<int>(rtp_rtx.fec.TotalBytes() *
- 8 / elapsed_sec / 1000));
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.FecBitrateSentInKbps",
+ static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
+ 1000));
}
}
} else if (vie_receiver_.GetRemoteSsrc() > 0) {
@@ -278,14 +282,18 @@
GetReceiveRtcpPacketTypeCounter(&rtcp_counter);
int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.NackPacketsSentPerMinute",
rtcp_counter.nack_packets * 60 / elapsed_sec);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.FirPacketsSentPerMinute",
rtcp_counter.fir_packets * 60 / elapsed_sec);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.PliPacketsSentPerMinute",
rtcp_counter.pli_packets * 60 / elapsed_sec);
if (rtcp_counter.nack_requests > 0) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+ "WebRTC.Video.UniqueNackRequestsSentInPercent",
rtcp_counter.UniqueNackRequestsInPercent());
}
}
@@ -297,32 +305,33 @@
rtp_rtx.Add(rtx);
elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.BitrateReceivedInKbps",
static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.MediaBitrateReceivedInKbps",
static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.PaddingBitrateReceivedInKbps",
static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
1000));
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps",
static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 /
elapsed_sec / 1000));
uint32_t ssrc = 0;
if (vie_receiver_.GetRtxSsrc(&ssrc)) {
- RTC_HISTOGRAM_COUNTS_10000(
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.RtxBitrateReceivedInKbps",
static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
1000));
}
if (vie_receiver_.IsFecEnabled()) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps",
- static_cast<int>(rtp_rtx.fec.TotalBytes() *
- 8 / elapsed_sec / 1000));
+ RTC_HISTOGRAM_COUNTS_SPARSE_10000(
+ "WebRTC.Video.FecBitrateReceivedInKbps",
+ static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec /
+ 1000));
}
}
}
diff --git a/webrtc/video/vie_receiver.cc b/webrtc/video/vie_receiver.cc
index 98c8c5d..4fb706c 100644
--- a/webrtc/video/vie_receiver.cc
+++ b/webrtc/video/vie_receiver.cc
@@ -68,14 +68,15 @@
void ViEReceiver::UpdateHistograms() {
FecPacketCounter counter = fec_receiver_->GetPacketCounter();
if (counter.num_packets > 0) {
- RTC_HISTOGRAM_PERCENTAGE(
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
"WebRTC.Video.ReceivedFecPacketsInPercent",
static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
}
if (counter.num_fec_packets > 0) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
- static_cast<int>(counter.num_recovered_packets *
- 100 / counter.num_fec_packets));
+ RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+ "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+ static_cast<int>(counter.num_recovered_packets * 100 /
+ counter.num_fec_packets));
}
}
