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

webrtc/video/rtp_stream_receiver.cc

 /*
  *  Copyright (c) 2012 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 602 matching lines @@
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 
 void RtpStreamReceiver::UpdateHistograms() {
   FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter();
+  if (counter.first_packet_time_ms == -1)
+    return;
+
+  int64_t elapsed_sec =
+      (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+    return;
+
   if (counter.num_packets > 0) {
     RTC_HISTOGRAM_PERCENTAGE(
         "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));
   }
 }
 
 void RtpStreamReceiver::EnableReceiveRtpHeaderExtension(
     const std::string& extension, int id) {
   // One-byte-extension local identifiers are in the range 1-14 inclusive.
   RTC_DCHECK_GE(id, 1);
   RTC_DCHECK_LE(id, 14);
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
 
 }  // namespace webrtc