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webrtc/video/receive_statistics_proxy.cc
Mirko Bonadei 675513b96a Stop using LOG macros in favor of RTC_ prefixed macros. 
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00

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32 KiB
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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "video/receive_statistics_proxy.h"
#include <algorithm>
#include <cmath>
#include <sstream>
#include <utility>
#include "modules/pacing/alr_detector.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
// Periodic time interval for processing samples for |freq_offset_counter_|.
const int64_t kFreqOffsetProcessIntervalMs = 40000;
// Configuration for bad call detection.
const int kBadCallMinRequiredSamples = 10;
const int kMinSampleLengthMs = 990;
const int kNumMeasurements = 10;
const int kNumMeasurementsVariance = kNumMeasurements * 1.5;
const float kBadFraction = 0.8f;
// For fps:
// Low means low enough to be bad, high means high enough to be good
const int kLowFpsThreshold = 12;
const int kHighFpsThreshold = 14;
// For qp and fps variance:
// Low means low enough to be good, high means high enough to be bad
const int kLowQpThresholdVp8 = 60;
const int kHighQpThresholdVp8 = 70;
const int kLowVarianceThreshold = 1;
const int kHighVarianceThreshold = 2;
// Some metrics are reported as a maximum over this period.
// This should be synchronized with a typical getStats polling interval in
// the clients.
const int kMovingMaxWindowMs = 1000;
// How large window we use to calculate the framerate/bitrate.
const int kRateStatisticsWindowSizeMs = 1000;
// Some sane ballpark estimate for maximum common value of inter-frame delay.
// Values below that will be stored explicitly in the array,
// values above - in the map.
const int kMaxCommonInterframeDelayMs = 500;
std::string UmaPrefixForContentType(VideoContentType content_type) {
std::stringstream ss;
ss << "WebRTC.Video";
if (videocontenttypehelpers::IsScreenshare(content_type)) {
ss << ".Screenshare";
}
return ss.str();
}
std::string UmaSuffixForContentType(VideoContentType content_type) {
std::stringstream ss;
int simulcast_id = videocontenttypehelpers::GetSimulcastId(content_type);
if (simulcast_id > 0) {
ss << ".S" << simulcast_id - 1;
}
int experiment_id = videocontenttypehelpers::GetExperimentId(content_type);
if (experiment_id > 0) {
ss << ".ExperimentGroup" << experiment_id - 1;
}
return ss.str();
}
} // namespace
ReceiveStatisticsProxy::ReceiveStatisticsProxy(
const VideoReceiveStream::Config* config,
Clock* clock)
: clock_(clock),
config_(*config),
start_ms_(clock->TimeInMilliseconds()),
last_sample_time_(clock->TimeInMilliseconds()),
fps_threshold_(kLowFpsThreshold,
kHighFpsThreshold,
kBadFraction,
kNumMeasurements),
qp_threshold_(kLowQpThresholdVp8,
kHighQpThresholdVp8,
kBadFraction,
kNumMeasurements),
variance_threshold_(kLowVarianceThreshold,
kHighVarianceThreshold,
kBadFraction,
kNumMeasurementsVariance),
num_bad_states_(0),
num_certain_states_(0),
// 1000ms window, scale 1000 for ms to s.
decode_fps_estimator_(1000, 1000),
renders_fps_estimator_(1000, 1000),
render_fps_tracker_(100, 10u),
render_pixel_tracker_(100, 10u),
total_byte_tracker_(100, 10u), // bucket_interval_ms, bucket_count
interframe_delay_max_moving_(kMovingMaxWindowMs),
freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
first_report_block_time_ms_(-1),
avg_rtt_ms_(0),
last_content_type_(VideoContentType::UNSPECIFIED),
timing_frame_info_counter_(kMovingMaxWindowMs) {
stats_.ssrc = config_.rtp.remote_ssrc;
// TODO(brandtr): Replace |rtx_stats_| with a single instance of
// StreamDataCounters.
if (config_.rtp.rtx_ssrc) {
rtx_stats_[config_.rtp.rtx_ssrc] = StreamDataCounters();
}
}
ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
UpdateHistograms();
}
void ReceiveStatisticsProxy::UpdateHistograms() {
int stream_duration_sec = (clock_->TimeInMilliseconds() - start_ms_) / 1000;
if (stats_.frame_counts.key_frames > 0 ||
stats_.frame_counts.delta_frames > 0) {
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.ReceiveStreamLifetimeInSeconds",
stream_duration_sec);
RTC_LOG(LS_INFO) << "WebRTC.Video.ReceiveStreamLifetimeInSeconds "
<< stream_duration_sec;
}
if (first_report_block_time_ms_ != -1 &&
((clock_->TimeInMilliseconds() - first_report_block_time_ms_) / 1000) >=
metrics::kMinRunTimeInSeconds) {
int fraction_lost = report_block_stats_.FractionLostInPercent();
if (fraction_lost != -1) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
fraction_lost);
RTC_LOG(LS_INFO) << "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",
round(render_fps_tracker_.ComputeTotalRate()));
RTC_HISTOGRAM_COUNTS_100000(
"WebRTC.Video.RenderSqrtPixelsPerSecond",
round(render_pixel_tracker_.ComputeTotalRate()));
}
int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
if (sync_offset_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
RTC_LOG(LS_INFO) << "WebRTC.Video.AVSyncOffsetInMs " << sync_offset_ms;
}
AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
if (freq_offset_stats.num_samples > 0) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
freq_offset_stats.average);
RTC_LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
<< freq_offset_stats.ToString();
}
int num_total_frames =
stats_.frame_counts.key_frames + stats_.frame_counts.delta_frames;
if (num_total_frames >= kMinRequiredSamples) {
int num_key_frames = stats_.frame_counts.key_frames;
int key_frames_permille =
(num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
key_frames_permille);
RTC_LOG(LS_INFO) << "WebRTC.Video.KeyFramesReceivedInPermille "
<< key_frames_permille;
}
int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
if (qp != -1) {
RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
RTC_LOG(LS_INFO) << "WebRTC.Video.Decoded.Vp8.Qp " << qp;
}
int decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
if (decode_ms != -1) {
RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
RTC_LOG(LS_INFO) << "WebRTC.Video.DecodeTimeInMs " << decode_ms;
}
int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredSamples);
if (jb_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
jb_delay_ms);
RTC_LOG(LS_INFO) << "WebRTC.Video.JitterBufferDelayInMs " << jb_delay_ms;
}
int target_delay_ms = target_delay_counter_.Avg(kMinRequiredSamples);
if (target_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
RTC_LOG(LS_INFO) << "WebRTC.Video.TargetDelayInMs " << target_delay_ms;
}
int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
if (current_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
current_delay_ms);
RTC_LOG(LS_INFO) << "WebRTC.Video.CurrentDelayInMs " << current_delay_ms;
}
int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
if (delay_ms != -1)
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
// Aggregate content_specific_stats_ by removing experiment or simulcast
// information;
std::map<VideoContentType, ContentSpecificStats> aggregated_stats;
for (auto it : content_specific_stats_) {
// Calculate simulcast specific metrics (".S0" ... ".S2" suffixes).
VideoContentType content_type = it.first;
if (videocontenttypehelpers::GetSimulcastId(content_type) > 0) {
// Aggregate on experiment id.
videocontenttypehelpers::SetExperimentId(&content_type, 0);
aggregated_stats[content_type].Add(it.second);
}
// Calculate experiment specific metrics (".ExperimentGroup[0-7]" suffixes).
content_type = it.first;
if (videocontenttypehelpers::GetExperimentId(content_type) > 0) {
// Aggregate on simulcast id.
videocontenttypehelpers::SetSimulcastId(&content_type, 0);
aggregated_stats[content_type].Add(it.second);
}
// Calculate aggregated metrics (no suffixes. Aggregated on everything).
content_type = it.first;
videocontenttypehelpers::SetSimulcastId(&content_type, 0);
videocontenttypehelpers::SetExperimentId(&content_type, 0);
aggregated_stats[content_type].Add(it.second);
}
for (auto it : aggregated_stats) {
// For the metric Foo we report the following slices:
// WebRTC.Video.Foo,
// WebRTC.Video.Screenshare.Foo,
// WebRTC.Video.Foo.S[0-3],
// WebRTC.Video.Foo.ExperimentGroup[0-7],
// WebRTC.Video.Screenshare.Foo.S[0-3],
// WebRTC.Video.Screenshare.Foo.ExperimentGroup[0-7].
auto content_type = it.first;
auto stats = it.second;
std::string uma_prefix = UmaPrefixForContentType(content_type);
std::string uma_suffix = UmaSuffixForContentType(content_type);
// Metrics can be sliced on either simulcast id or experiment id but not
// both.
RTC_DCHECK(videocontenttypehelpers::GetExperimentId(content_type) == 0 ||
videocontenttypehelpers::GetSimulcastId(content_type) == 0);
int e2e_delay_ms = stats.e2e_delay_counter.Avg(kMinRequiredSamples);
if (e2e_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".EndToEndDelayInMs" + uma_suffix, e2e_delay_ms);
RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix
<< " " << e2e_delay_ms;
}
int e2e_delay_max_ms = stats.e2e_delay_counter.Max();
if (e2e_delay_max_ms != -1 && e2e_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_100000(
uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, e2e_delay_max_ms);
RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix
<< " " << e2e_delay_max_ms;
}
int interframe_delay_ms =
stats.interframe_delay_counter.Avg(kMinRequiredSamples);
if (interframe_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".InterframeDelayInMs" + uma_suffix,
interframe_delay_ms);
RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix
<< " " << interframe_delay_ms;
}
int interframe_delay_max_ms = stats.interframe_delay_counter.Max();
if (interframe_delay_max_ms != -1 && interframe_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix,
interframe_delay_max_ms);
RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix
<< " " << interframe_delay_max_ms;
}
rtc::Optional<uint32_t> interframe_delay_95p_ms =
stats.interframe_delay_percentiles.GetPercentile(0.95f);
if (interframe_delay_95p_ms && interframe_delay_ms != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".InterframeDelay95PercentileInMs" + uma_suffix,
*interframe_delay_95p_ms);
RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelay95PercentileInMs"
<< uma_suffix << " " << *interframe_delay_95p_ms;
}
int width = stats.received_width.Avg(kMinRequiredSamples);
if (width != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, width);
RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix
<< " " << width;
}
int height = stats.received_height.Avg(kMinRequiredSamples);
if (height != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, height);
RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix
<< " " << height;
}
if (content_type != VideoContentType::UNSPECIFIED) {
// Don't report these 3 metrics unsliced, as more precise variants
// are reported separately in this method.
float flow_duration_sec = stats.flow_duration_ms / 1000.0;
if (flow_duration_sec >= metrics::kMinRunTimeInSeconds) {
int media_bitrate_kbps = static_cast<int>(stats.total_media_bytes * 8 /
flow_duration_sec / 1000);
RTC_HISTOGRAM_COUNTS_SPARSE_10000(
uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix,
media_bitrate_kbps);
RTC_LOG(LS_INFO) << uma_prefix << ".MediaBitrateReceivedInKbps"
<< uma_suffix << " " << media_bitrate_kbps;
}
int num_total_frames =
stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
if (num_total_frames >= kMinRequiredSamples) {
int num_key_frames = stats.frame_counts.key_frames;
int key_frames_permille =
(num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
RTC_HISTOGRAM_COUNTS_SPARSE_1000(
uma_prefix + ".KeyFramesReceivedInPermille" + uma_suffix,
key_frames_permille);
RTC_LOG(LS_INFO) << uma_prefix << ".KeyFramesReceivedInPermille"
<< uma_suffix << " " << key_frames_permille;
}
int qp = stats.qp_counter.Avg(kMinRequiredSamples);
if (qp != -1) {
RTC_HISTOGRAM_COUNTS_SPARSE_200(
uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, qp);
RTC_LOG(LS_INFO) << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " "
<< qp;
}
}
}
StreamDataCounters rtp = stats_.rtp_stats;
StreamDataCounters rtx;
for (auto it : rtx_stats_)
rtx.Add(it.second);
StreamDataCounters rtp_rtx = rtp;
rtp_rtx.Add(rtx);
int64_t elapsed_sec =
rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.BitrateReceivedInKbps",
static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
1000));
int media_bitrate_kbs =
static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
media_bitrate_kbs);
RTC_LOG(LS_INFO) << "WebRTC.Video.MediaBitrateReceivedInKbps "
<< media_bitrate_kbs;
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.PaddingBitrateReceivedInKbps",
static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
1000));
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps",
static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
1000));
if (!rtx_stats_.empty()) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
static_cast<int>(rtx.transmitted.TotalBytes() *
8 / elapsed_sec / 1000));
}
if (config_.rtp.ulpfec_payload_type != -1) {
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.Video.FecBitrateReceivedInKbps",
static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
}
const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts;
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
counters.nack_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
counters.fir_packets * 60 / elapsed_sec);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
counters.pli_packets * 60 / elapsed_sec);
if (counters.nack_requests > 0) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
counters.UniqueNackRequestsInPercent());
}
}
if (num_certain_states_ >= kBadCallMinRequiredSamples) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Any",
100 * num_bad_states_ / num_certain_states_);
}
rtc::Optional<double> fps_fraction =
fps_threshold_.FractionHigh(kBadCallMinRequiredSamples);
if (fps_fraction) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRate",
static_cast<int>(100 * (1 - *fps_fraction)));
}
rtc::Optional<double> variance_fraction =
variance_threshold_.FractionHigh(kBadCallMinRequiredSamples);
if (variance_fraction) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRateVariance",
static_cast<int>(100 * *variance_fraction));
}
rtc::Optional<double> qp_fraction =
qp_threshold_.FractionHigh(kBadCallMinRequiredSamples);
if (qp_fraction) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Qp",
static_cast<int>(100 * *qp_fraction));
}
}
void ReceiveStatisticsProxy::QualitySample() {
int64_t now = clock_->TimeInMilliseconds();
if (last_sample_time_ + kMinSampleLengthMs > now)
return;
double fps =
render_fps_tracker_.ComputeRateForInterval(now - last_sample_time_);
int qp = qp_sample_.Avg(1);
bool prev_fps_bad = !fps_threshold_.IsHigh().value_or(true);
bool prev_qp_bad = qp_threshold_.IsHigh().value_or(false);
bool prev_variance_bad = variance_threshold_.IsHigh().value_or(false);
bool prev_any_bad = prev_fps_bad || prev_qp_bad || prev_variance_bad;
fps_threshold_.AddMeasurement(static_cast<int>(fps));
if (qp != -1)
qp_threshold_.AddMeasurement(qp);
rtc::Optional<double> fps_variance_opt = fps_threshold_.CalculateVariance();
double fps_variance = fps_variance_opt.value_or(0);
if (fps_variance_opt) {
variance_threshold_.AddMeasurement(static_cast<int>(fps_variance));
}
bool fps_bad = !fps_threshold_.IsHigh().value_or(true);
bool qp_bad = qp_threshold_.IsHigh().value_or(false);
bool variance_bad = variance_threshold_.IsHigh().value_or(false);
bool any_bad = fps_bad || qp_bad || variance_bad;
if (!prev_any_bad && any_bad) {
RTC_LOG(LS_INFO) << "Bad call (any) start: " << now;
} else if (prev_any_bad && !any_bad) {
RTC_LOG(LS_INFO) << "Bad call (any) end: " << now;
}
if (!prev_fps_bad && fps_bad) {
RTC_LOG(LS_INFO) << "Bad call (fps) start: " << now;
} else if (prev_fps_bad && !fps_bad) {
RTC_LOG(LS_INFO) << "Bad call (fps) end: " << now;
}
if (!prev_qp_bad && qp_bad) {
RTC_LOG(LS_INFO) << "Bad call (qp) start: " << now;
} else if (prev_qp_bad && !qp_bad) {
RTC_LOG(LS_INFO) << "Bad call (qp) end: " << now;
}
if (!prev_variance_bad && variance_bad) {
RTC_LOG(LS_INFO) << "Bad call (variance) start: " << now;
} else if (prev_variance_bad && !variance_bad) {
RTC_LOG(LS_INFO) << "Bad call (variance) end: " << now;
}
RTC_LOG(LS_VERBOSE) << "SAMPLE: sample_length: " << (now - last_sample_time_)
<< " fps: " << fps << " fps_bad: " << fps_bad
<< " qp: " << qp << " qp_bad: " << qp_bad
<< " variance_bad: " << variance_bad
<< " fps_variance: " << fps_variance;
last_sample_time_ = now;
qp_sample_.Reset();
if (fps_threshold_.IsHigh() || variance_threshold_.IsHigh() ||
qp_threshold_.IsHigh()) {
if (any_bad)
++num_bad_states_;
++num_certain_states_;
}
}
void ReceiveStatisticsProxy::UpdateFramerate(int64_t now_ms) const {
int64_t old_frames_ms = now_ms - kRateStatisticsWindowSizeMs;
while (!frame_window_.empty() &&
frame_window_.begin()->first < old_frames_ms) {
frame_window_.erase(frame_window_.begin());
}
size_t framerate =
(frame_window_.size() * 1000 + 500) / kRateStatisticsWindowSizeMs;
stats_.network_frame_rate = static_cast<int>(framerate);
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
rtc::CritScope lock(&crit_);
// Get current frame rates here, as only updating them on new frames prevents
// us from ever correctly displaying frame rate of 0.
int64_t now_ms = clock_->TimeInMilliseconds();
UpdateFramerate(now_ms);
stats_.render_frame_rate = renders_fps_estimator_.Rate(now_ms).value_or(0);
stats_.decode_frame_rate = decode_fps_estimator_.Rate(now_ms).value_or(0);
stats_.total_bitrate_bps =
static_cast<int>(total_byte_tracker_.ComputeRate() * 8);
stats_.interframe_delay_max_ms =
interframe_delay_max_moving_.Max(now_ms).value_or(-1);
stats_.timing_frame_info = timing_frame_info_counter_.Max(now_ms);
stats_.content_type = last_content_type_;
return stats_;
}
void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
rtc::CritScope lock(&crit_);
stats_.current_payload_type = payload_type;
}
void ReceiveStatisticsProxy::OnDecoderImplementationName(
const char* implementation_name) {
rtc::CritScope lock(&crit_);
stats_.decoder_implementation_name = implementation_name;
}
void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
unsigned int bitrate_bps) {
rtc::CritScope lock(&crit_);
if (stats_.rtp_stats.first_packet_time_ms != -1)
QualitySample();
}
void ReceiveStatisticsProxy::OnFrameBufferTimingsUpdated(
int decode_ms,
int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
int jitter_buffer_ms,
int min_playout_delay_ms,
int render_delay_ms) {
rtc::CritScope lock(&crit_);
stats_.decode_ms = decode_ms;
stats_.max_decode_ms = max_decode_ms;
stats_.current_delay_ms = current_delay_ms;
stats_.target_delay_ms = target_delay_ms;
stats_.jitter_buffer_ms = jitter_buffer_ms;
stats_.min_playout_delay_ms = min_playout_delay_ms;
stats_.render_delay_ms = render_delay_ms;
decode_time_counter_.Add(decode_ms);
jitter_buffer_delay_counter_.Add(jitter_buffer_ms);
target_delay_counter_.Add(target_delay_ms);
current_delay_counter_.Add(current_delay_ms);
// Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
// render delay).
delay_counter_.Add(target_delay_ms + avg_rtt_ms_ / 2);
}
void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated(
const TimingFrameInfo& info) {
rtc::CritScope lock(&crit_);
int64_t now_ms = clock_->TimeInMilliseconds();
timing_frame_info_counter_.Add(info, now_ms);
}
void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
uint32_t ssrc,
const RtcpPacketTypeCounter& packet_counter) {
rtc::CritScope lock(&crit_);
if (stats_.ssrc != ssrc)
return;
stats_.rtcp_packet_type_counts = packet_counter;
}
void ReceiveStatisticsProxy::StatisticsUpdated(
const webrtc::RtcpStatistics& statistics,
uint32_t ssrc) {
rtc::CritScope lock(&crit_);
// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
// receive stats from one of them.
if (stats_.ssrc != ssrc)
return;
stats_.rtcp_stats = statistics;
report_block_stats_.Store(statistics, ssrc, 0);
if (first_report_block_time_ms_ == -1)
first_report_block_time_ms_ = clock_->TimeInMilliseconds();
}
void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
rtc::CritScope lock(&crit_);
// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
// receive stats from one of them.
if (stats_.ssrc != ssrc)
return;
stats_.c_name = cname;
}
void ReceiveStatisticsProxy::DataCountersUpdated(
const webrtc::StreamDataCounters& counters,
uint32_t ssrc) {
size_t last_total_bytes = 0;
size_t total_bytes = 0;
rtc::CritScope lock(&crit_);
if (ssrc == stats_.ssrc) {
last_total_bytes = stats_.rtp_stats.transmitted.TotalBytes();
total_bytes = counters.transmitted.TotalBytes();
stats_.rtp_stats = counters;
} else {
auto it = rtx_stats_.find(ssrc);
if (it != rtx_stats_.end()) {
last_total_bytes = it->second.transmitted.TotalBytes();
total_bytes = counters.transmitted.TotalBytes();
it->second = counters;
} else {
RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
}
}
if (total_bytes > last_total_bytes)
total_byte_tracker_.AddSamples(total_bytes - last_total_bytes);
}
void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp,
VideoContentType content_type) {
rtc::CritScope lock(&crit_);
uint64_t now = clock_->TimeInMilliseconds();
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[content_type];
++stats_.frames_decoded;
if (qp) {
if (!stats_.qp_sum) {
if (stats_.frames_decoded != 1) {
RTC_LOG(LS_WARNING)
<< "Frames decoded was not 1 when first qp value was received.";
stats_.frames_decoded = 1;
}
stats_.qp_sum = rtc::Optional<uint64_t>(0);
}
*stats_.qp_sum += *qp;
content_specific_stats->qp_counter.Add(*qp);
} else if (stats_.qp_sum) {
RTC_LOG(LS_WARNING)
<< "QP sum was already set and no QP was given for a frame.";
stats_.qp_sum = rtc::Optional<uint64_t>();
}
last_content_type_ = content_type;
decode_fps_estimator_.Update(1, now);
if (last_decoded_frame_time_ms_) {
int64_t interframe_delay_ms = now - *last_decoded_frame_time_ms_;
RTC_DCHECK_GE(interframe_delay_ms, 0);
interframe_delay_max_moving_.Add(interframe_delay_ms, now);
content_specific_stats->interframe_delay_counter.Add(interframe_delay_ms);
content_specific_stats->interframe_delay_percentiles.Add(
interframe_delay_ms);
content_specific_stats->flow_duration_ms += interframe_delay_ms;
}
last_decoded_frame_time_ms_.emplace(now);
}
void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
int width = frame.width();
int height = frame.height();
RTC_DCHECK_GT(width, 0);
RTC_DCHECK_GT(height, 0);
uint64_t now = clock_->TimeInMilliseconds();
rtc::CritScope lock(&crit_);
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[last_content_type_];
renders_fps_estimator_.Update(1, now);
++stats_.frames_rendered;
stats_.width = width;
stats_.height = height;
render_fps_tracker_.AddSamples(1);
render_pixel_tracker_.AddSamples(sqrt(width * height));
content_specific_stats->received_width.Add(width);
content_specific_stats->received_height.Add(height);
if (frame.ntp_time_ms() > 0) {
int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
if (delay_ms >= 0) {
content_specific_stats->e2e_delay_counter.Add(delay_ms);
}
}
}
void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms,
double estimated_freq_khz) {
rtc::CritScope lock(&crit_);
sync_offset_counter_.Add(std::abs(sync_offset_ms));
stats_.sync_offset_ms = sync_offset_ms;
const double kMaxFreqKhz = 10000.0;
int offset_khz = kMaxFreqKhz;
// Should not be zero or negative. If so, report max.
if (estimated_freq_khz < kMaxFreqKhz && estimated_freq_khz > 0.0)
offset_khz = static_cast<int>(std::fabs(estimated_freq_khz - 90.0) + 0.5);
freq_offset_counter_.Add(offset_khz);
}
void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
uint32_t frameRate) {
}
void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
size_t size_bytes,
VideoContentType content_type) {
rtc::CritScope lock(&crit_);
if (is_keyframe) {
++stats_.frame_counts.key_frames;
} else {
++stats_.frame_counts.delta_frames;
}
ContentSpecificStats* content_specific_stats =
&content_specific_stats_[content_type];
content_specific_stats->total_media_bytes += size_bytes;
if (is_keyframe) {
++content_specific_stats->frame_counts.key_frames;
} else {
++content_specific_stats->frame_counts.delta_frames;
}
int64_t now_ms = clock_->TimeInMilliseconds();
frame_window_.insert(std::make_pair(now_ms, size_bytes));
UpdateFramerate(now_ms);
}
void ReceiveStatisticsProxy::OnFrameCountsUpdated(
const FrameCounts& frame_counts) {
rtc::CritScope lock(&crit_);
stats_.frame_counts = frame_counts;
}
void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
rtc::CritScope lock(&crit_);
stats_.discarded_packets = discarded_packets;
}
void ReceiveStatisticsProxy::OnPreDecode(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) {
if (!codec_specific_info || encoded_image.qp_ == -1) {
return;
}
if (codec_specific_info->codecType == kVideoCodecVP8) {
qp_counters_.vp8.Add(encoded_image.qp_);
rtc::CritScope lock(&crit_);
qp_sample_.Add(encoded_image.qp_);
}
}
void ReceiveStatisticsProxy::OnStreamInactive() {
// TODO(sprang): Figure out any other state that should be reset.
rtc::CritScope lock(&crit_);
// Don't report inter-frame delay if stream was paused.
last_decoded_frame_time_ms_.reset();
}
void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
sum += sample;
++num_samples;
if (!max || sample > *max) {
max.emplace(sample);
}
}
void ReceiveStatisticsProxy::SampleCounter::Add(const SampleCounter& other) {
sum += other.sum;
num_samples += other.num_samples;
if (other.max && (!max || *max < *other.max))
max = other.max;
}
int ReceiveStatisticsProxy::SampleCounter::Avg(
int64_t min_required_samples) const {
if (num_samples < min_required_samples || num_samples == 0)
return -1;
return static_cast<int>(sum / num_samples);
}
int ReceiveStatisticsProxy::SampleCounter::Max() const {
return max.value_or(-1);
}
void ReceiveStatisticsProxy::SampleCounter::Reset() {
num_samples = 0;
sum = 0;
max.reset();
}
void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
int64_t max_rtt_ms) {
rtc::CritScope lock(&crit_);
avg_rtt_ms_ = avg_rtt_ms;
}
ReceiveStatisticsProxy::ContentSpecificStats::ContentSpecificStats()
: interframe_delay_percentiles(kMaxCommonInterframeDelayMs) {}
void ReceiveStatisticsProxy::ContentSpecificStats::Add(
const ContentSpecificStats& other) {
e2e_delay_counter.Add(other.e2e_delay_counter);
interframe_delay_counter.Add(other.interframe_delay_counter);
flow_duration_ms += other.flow_duration_ms;
total_media_bytes += other.total_media_bytes;
received_height.Add(other.received_height);
received_width.Add(other.received_width);
qp_counter.Add(other.qp_counter);
frame_counts.key_frames += other.frame_counts.key_frames;
frame_counts.delta_frames += other.frame_counts.delta_frames;
interframe_delay_percentiles.Add(other.interframe_delay_percentiles);
}
} // namespace webrtc
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