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Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.

Browse Source 
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}
This commit is contained in:
asapersson
2015-12-21 01:46:20 -08:00
committed by Commit bot
parent 9fca7e18c3
commit 53805324c0
12 changed files with 149 additions and 130 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
webrtc/call/call.cc
View File

@ -253,11 +253,11 @@ 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",
RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
send_bitrate_kbps);
}
if (pacer_bitrate_kbps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
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",
RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
video_bitrate_kbps);
}
if (audio_bitrate_kbps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
audio_bitrate_kbps);
}
if (rtcp_bitrate_bps > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
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);
}

2
webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
View File

@ -97,7 +97,7 @@ void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
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);
}
}

4
webrtc/modules/audio_coding/neteq/statistics_calculator.cc
View File

@ -50,7 +50,7 @@ void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
}
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(
@ -187,7 +187,7 @@ void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
}
void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.DelayedPacketOutageEventMs",
outage_duration_ms, 1 /* min */, 2000 /* max */,
100 /* bucket count */);
delayed_packet_outage_counter_.RegisterSample();

15
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -1346,8 +1346,9 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
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 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
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 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
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);
}
@ -1390,7 +1391,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
capture_.last_stream_delay_ms = 0;
if (capture_.aec_system_delay_jumps > -1) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
capture_.aec_system_delay_jumps, 51);
}
capture_.aec_system_delay_jumps = -1;

21
webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
View File

@ -146,7 +146,7 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
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,
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 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
} 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);
}
}

9
webrtc/modules/video_coding/jitter_buffer.cc
View File

@ -281,17 +281,18 @@ void VCMJitterBuffer::UpdateHistograms() {
return;
}
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DiscardedPacketsInPercent",
num_discarded_packets_ * 100 / num_packets_);
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
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));

8
webrtc/modules/video_coding/timing.cc
View File

@ -62,12 +62,14 @@ void VCMTiming::UpdateHistograms() const {
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_);
}

35
webrtc/system_wrappers/include/metrics.h
View File

@ -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, \
#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, \
#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, \
#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 {

18
webrtc/video/receive_statistics_proxy.cc
View File

@ -37,37 +37,39 @@ ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
void ReceiveStatisticsProxy::UpdateHistograms() {
int fraction_lost = report_block_stats_.FractionLostInPercent();
if (fraction_lost != -1) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
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 {

39
webrtc/video/send_statistics_proxy.cc
View File

@ -57,7 +57,7 @@ HistogramCodecType PayloadNameToHistogramCodecType(
}
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
@ -98,55 +98,62 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms() {
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",
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",
RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix_ + "SendSideDelayMaxInMs",
max_delay_ms);
}
}

61
webrtc/video/vie_channel.cc
View File

@ -202,7 +202,7 @@ void ViEChannel::UpdateHistograms() {
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);
}
}
@ -212,20 +212,23 @@ void ViEChannel::UpdateHistograms() {
GetSendRtcpPacketTypeCounter(&rtcp_counter);
int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000;
if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.NackPacketsReceivedPerMinute",
rtcp_counter.nack_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
"WebRTC.Video.FirPacketsReceivedPerMinute",
rtcp_counter.fir_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
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",
RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.SentPacketsLostInPercent",
fraction_lost);
}
}
@ -239,23 +242,23 @@ void ViEChannel::UpdateHistograms() {
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 @@ void ViEChannel::UpdateHistograms() {
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 @@ void ViEChannel::UpdateHistograms() {
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 @@ void ViEChannel::UpdateHistograms() {
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));
}
}
}

9
webrtc/video/vie_receiver.cc
View File

@ -68,14 +68,15 @@ ViEReceiver::~ViEReceiver() {
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));
}
}