Update video histograms that do not have a minimum lifetime limit before being recorded.

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.
  */
 
@@ skipping 16 matching lines @@
     const VideoReceiveStream::Config* config,
     Clock* clock)
     : clock_(clock),
       config_(*config),
       start_ms_(clock->TimeInMilliseconds()),
       // 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) {
+      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);
 
-  int fraction_lost = report_block_stats_.FractionLostInPercent();
-  if (fraction_lost != -1) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
-                             fraction_lost);
+  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);
@@ skipping 154 matching lines @@
 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;
 }
@@ skipping 93 matching lines @@
   ++num_samples;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc