video/receive_statistics_proxy.cc - src/webrtc - Git at Google

 /*
  *  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 "webrtc/video/receive_statistics_proxy.h"

 #include <cmath>

 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/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;
 }  // 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),
       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
       first_report_block_time_ms_(-1) {
   stats_.ssrc = config_.rtp.remote_ssrc;
   for (auto it : config_.rtp.rtx)
     rtx_stats_[it.second.ssrc] = StreamDataCounters();
 }

 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
 }

 void ReceiveStatisticsProxy::UpdateHistograms() {
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);

   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);
     }
   }

   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 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);
   }
   int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
   if (sync_offset_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("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);
     LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
                  << freq_offset_stats.ToString();
   }

   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
   if (qp != -1)
     RTC_HISTOGRAM_COUNTS_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);

   if (field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") !=
         "Enabled") {
     int jb_delay_ms =
         jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
     if (jb_delay_ms != -1) {
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
                                  jb_delay_ms);
     }
   }
   int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (target_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
   }
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);

   int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
   if (e2e_delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);

   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));
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.MediaBitrateReceivedInKbps",
         static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
     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.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) {
     LOG(LS_INFO) << "Bad call (any) start: " << now;
   } else if (prev_any_bad && !any_bad) {
     LOG(LS_INFO) << "Bad call (any) end: " << now;
   }

   if (!prev_fps_bad && fps_bad) {
     LOG(LS_INFO) << "Bad call (fps) start: " << now;
   } else if (prev_fps_bad && !fps_bad) {
     LOG(LS_INFO) << "Bad call (fps) end: " << now;
   }

   if (!prev_qp_bad && qp_bad) {
     LOG(LS_INFO) << "Bad call (qp) start: " << now;
   } else if (prev_qp_bad && !qp_bad) {
     LOG(LS_INFO) << "Bad call (qp) end: " << now;
   }

   if (!prev_variance_bad && variance_bad) {
     LOG(LS_INFO) << "Bad call (variance) start: " << now;
   } else if (prev_variance_bad && !variance_bad) {
     LOG(LS_INFO) << "Bad call (variance) end: " << now;
   }

   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_;
   }
 }

 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
   rtc::CritScope lock(&crit_);
   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();
   stats_.network_frame_rate = framerate;
   stats_.total_bitrate_bps = bitrate_bps;
 }

 void ReceiveStatisticsProxy::OnDecoderTiming(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,
                                              int64_t rtt_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 + rtt_ms / 2);
 }

 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) {
   rtc::CritScope lock(&crit_);
   if (ssrc == stats_.ssrc) {
     stats_.rtp_stats = counters;
   } else {
     auto it = rtx_stats_.find(ssrc);
     if (it != rtx_stats_.end()) {
       it->second = counters;
     } else {
       RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
     }
   }
 }

 void ReceiveStatisticsProxy::OnDecodedFrame() {
   uint64_t now = clock_->TimeInMilliseconds();

   rtc::CritScope lock(&crit_);
   ++stats_.frames_decoded;
   decode_fps_estimator_.Update(1, now);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now).value_or(0);
 }

 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_);
   renders_fps_estimator_.Update(1, now);
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now).value_or(0);
   stats_.width = width;
   stats_.height = height;
   render_width_counter_.Add(width);
   render_height_counter_.Add(height);
   render_fps_tracker_.AddSamples(1);
   render_pixel_tracker_.AddSamples(sqrt(width * height));

   if (frame.ntp_time_ms() > 0) {
     int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
     if (delay_ms >= 0)
       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::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::SampleCounter::Add(int sample) {
   sum += sample;
   ++num_samples;
 }

 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);
 }

 void ReceiveStatisticsProxy::SampleCounter::Reset() {
   num_samples = 0;
   sum = 0;
 }

 }  // namespace webrtc
	/*
	* 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 "webrtc/video/receive_statistics_proxy.h"

	#include <cmath>

	#include "webrtc/base/checks.h"
	#include "webrtc/base/logging.h"
	#include "webrtc/modules/video_coding/include/video_codec_interface.h"
	#include "webrtc/system_wrappers/include/clock.h"
	#include "webrtc/system_wrappers/include/field_trial.h"
	#include "webrtc/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;
	} // 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),
	freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
	first_report_block_time_ms_(-1) {
	stats_.ssrc = config_.rtp.remote_ssrc;
	for (auto it : config_.rtp.rtx)
	rtx_stats_[it.second.ssrc] = StreamDataCounters();
	}

	ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
	UpdateHistograms();
	}

	void ReceiveStatisticsProxy::UpdateHistograms() {
	RTC_HISTOGRAM_COUNTS_100000(
	"WebRTC.Video.ReceiveStreamLifetimeInSeconds",
	(clock_->TimeInMilliseconds() - start_ms_) / 1000);

	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);
	}
	}

	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 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);
	}
	int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
	if (sync_offset_ms != -1) {
	RTC_HISTOGRAM_COUNTS_10000("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);
	LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
	<< freq_offset_stats.ToString();
	}

	int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
	if (qp != -1)
	RTC_HISTOGRAM_COUNTS_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);

	if (field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") !=
	"Enabled") {
	int jb_delay_ms =
	jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
	if (jb_delay_ms != -1) {
	RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
	jb_delay_ms);
	}
	}
	int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
	if (target_delay_ms != -1) {
	RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
	}
	int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
	if (current_delay_ms != -1) {
	RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
	current_delay_ms);
	}
	int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
	if (delay_ms != -1)
	RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);

	int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
	if (e2e_delay_ms != -1)
	RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);

	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));
	RTC_HISTOGRAM_COUNTS_10000(
	"WebRTC.Video.MediaBitrateReceivedInKbps",
	static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
	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.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) {
	LOG(LS_INFO) << "Bad call (any) start: " << now;
	} else if (prev_any_bad && !any_bad) {
	LOG(LS_INFO) << "Bad call (any) end: " << now;
	}

	if (!prev_fps_bad && fps_bad) {
	LOG(LS_INFO) << "Bad call (fps) start: " << now;
	} else if (prev_fps_bad && !fps_bad) {
	LOG(LS_INFO) << "Bad call (fps) end: " << now;
	}

	if (!prev_qp_bad && qp_bad) {
	LOG(LS_INFO) << "Bad call (qp) start: " << now;
	} else if (prev_qp_bad && !qp_bad) {
	LOG(LS_INFO) << "Bad call (qp) end: " << now;
	}

	if (!prev_variance_bad && variance_bad) {
	LOG(LS_INFO) << "Bad call (variance) start: " << now;
	} else if (prev_variance_bad && !variance_bad) {
	LOG(LS_INFO) << "Bad call (variance) end: " << now;
	}

	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_;
	}
	}

	VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
	rtc::CritScope lock(&crit_);
	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();
	stats_.network_frame_rate = framerate;
	stats_.total_bitrate_bps = bitrate_bps;
	}

	void ReceiveStatisticsProxy::OnDecoderTiming(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,
	int64_t rtt_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 + rtt_ms / 2);
	}

	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) {
	rtc::CritScope lock(&crit_);
	if (ssrc == stats_.ssrc) {
	stats_.rtp_stats = counters;
	} else {
	auto it = rtx_stats_.find(ssrc);
	if (it != rtx_stats_.end()) {
	it->second = counters;
	} else {
	RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
	}
	}
	}

	void ReceiveStatisticsProxy::OnDecodedFrame() {
	uint64_t now = clock_->TimeInMilliseconds();

	rtc::CritScope lock(&crit_);
	++stats_.frames_decoded;
	decode_fps_estimator_.Update(1, now);
	stats_.decode_frame_rate = decode_fps_estimator_.Rate(now).value_or(0);
	}

	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_);
	renders_fps_estimator_.Update(1, now);
	stats_.render_frame_rate = renders_fps_estimator_.Rate(now).value_or(0);
	stats_.width = width;
	stats_.height = height;
	render_width_counter_.Add(width);
	render_height_counter_.Add(height);
	render_fps_tracker_.AddSamples(1);
	render_pixel_tracker_.AddSamples(sqrt(width * height));

	if (frame.ntp_time_ms() > 0) {
	int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
	if (delay_ms >= 0)
	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::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::SampleCounter::Add(int sample) {
	sum += sample;
	++num_samples;
	}

	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);
	}

	void ReceiveStatisticsProxy::SampleCounter::Reset() {
	num_samples = 0;
	sum = 0;
	}

	} // namespace webrtc