Add UMA stats to bad call detection.

webrtc/video/receive_statistics_proxy.cc

 /*
  *  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
@@ skipping 15 matching lines @@
                      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();
@@ skipping 121 matching lines @@
                                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);
@@ skipping 41 matching lines @@
     LOG(LS_WARNING) << "Bad call (variance) end: " << now;
   }
 
   LOG(LS_INFO) << "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_);
-  QualitySample();
+  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,
@@ skipping 151 matching lines @@
     return -1;
   return sum / num_samples;
 }
 
 void ReceiveStatisticsProxy::SampleCounter::Reset() {
   num_samples = 0;
   sum = 0;
 }
 
 }  // namespace webrtc