Only update codec type histogram if lifetime is long enough (10 sec).

webrtc/call/call.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 118 matching lines @@
     internal::AudioState* audio_state =
         static_cast<internal::AudioState*>(config_.audio_state.get());
     if (audio_state)
       return audio_state->voice_engine();
     else
       return nullptr;
   }
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
+  void UpdateHistograms();
   void UpdateAggregateNetworkState();
 
   Clock* const clock_;
 
   const int num_cpu_cores_;
   const std::unique_ptr<ProcessThread> module_process_thread_;
   const std::unique_ptr<ProcessThread> pacer_thread_;
   const std::unique_ptr<CallStats> call_stats_;
   const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
@@ skipping 40 matching lines @@
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
   uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
 
   std::map<std::string, rtc::NetworkRoute> network_routes_;
 
   VieRemb remb_;
   const std::unique_ptr<CongestionController> congestion_controller_;
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
+  const int64_t start_ms_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
@@ skipping 15 matching lines @@
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
       last_rtp_packet_received_ms_(-1),
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       min_allocated_send_bitrate_bps_(0),
       num_bitrate_updates_(0),
       configured_max_padding_bitrate_bps_(0),
-
       remb_(clock_),
       congestion_controller_(
           new CongestionController(clock_, this, &remb_, event_log_.get())),
-      video_send_delay_stats_(new SendDelayStats(clock_)) {
+      video_send_delay_stats_(new SendDelayStats(clock_)),
+      start_ms_(clock_->TimeInMilliseconds()) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
 
   Trace::CreateTrace();
@@ skipping 29 matching lines @@
       congestion_controller_->GetRemoteBitrateEstimator(true));
   module_process_thread_->DeRegisterModule(congestion_controller_.get());
   module_process_thread_->DeRegisterModule(call_stats_.get());
   module_process_thread_->Stop();
   call_stats_->DeregisterStatsObserver(congestion_controller_.get());
 
   // Only update histograms after process threads have been shut down, so that
   // they won't try to concurrently update stats.
   UpdateSendHistograms();
   UpdateReceiveHistograms();
+  UpdateHistograms();
 
   Trace::ReturnTrace();
 }
 
+void Call::UpdateHistograms() {
+  RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+      "WebRTC.Call.LifetimeInSeconds",
+      (clock_->TimeInMilliseconds() - start_ms_) / 1000);
+}
+
 void Call::UpdateSendHistograms() {
   if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   int send_bitrate_kbps =
       estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
   int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
@@ skipping 575 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc