Replace first_packet_sent_ms_ in Call.

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 199 matching lines @@
 
   void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
                                  MediaType media_type)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
   rtc::Optional<RtpPacketReceived> ParseRtpPacket(const uint8_t* packet,
                                                   size_t length,
                                                   const PacketTime& packet_time)
       SHARED_LOCKS_REQUIRED(receive_crit_);
 
-  void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
+  void UpdateSendHistograms(int64_t first_sent_packet_ms)
+      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_;
@@ skipping 53 matching lines @@
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
   webrtc::RtcEventLog* event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
-  int64_t first_packet_sent_ms_;
   RateCounter received_bytes_per_second_counter_;
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
@@ skipping 45 matching lines @@
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator(this)),
       config_(config),
       audio_network_state_(kNetworkDown),
       video_network_state_(kNetworkDown),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       event_log_(config.event_log),
-      first_packet_sent_ms_(-1),
       received_bytes_per_second_counter_(clock_, nullptr, true),
       received_audio_bytes_per_second_counter_(clock_, nullptr, true),
       received_video_bytes_per_second_counter_(clock_, nullptr, true),
       received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
       min_allocated_send_bitrate_bps_(0),
       configured_max_padding_bitrate_bps_(0),
       estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
       pacer_bitrate_kbps_counter_(clock_, nullptr, true),
       remb_(clock_),
       receive_side_cc_(clock_, &remb_, transport_send->packet_router()),
@@ skipping 48 matching lines @@
   pacer_thread_->DeRegisterModule(transport_send_->send_side_cc()->pacer());
   pacer_thread_->DeRegisterModule(
       receive_side_cc_.GetRemoteBitrateEstimator(true));
   module_process_thread_->DeRegisterModule(transport_send_->send_side_cc());
   module_process_thread_->DeRegisterModule(&receive_side_cc_);
   module_process_thread_->DeRegisterModule(call_stats_.get());
   module_process_thread_->Stop();
   call_stats_->DeregisterStatsObserver(&receive_side_cc_);
   call_stats_->DeregisterStatsObserver(transport_send_->send_side_cc());
 
+  int64_t first_sent_packet_ms =
+      transport_send_->send_side_cc()->GetFirstPacketTimeMs();
   // Only update histograms after process threads have been shut down, so that
   // they won't try to concurrently update stats.
   {
     rtc::CritScope lock(&bitrate_crit_);
-    UpdateSendHistograms();
+    UpdateSendHistograms(first_sent_packet_ms);
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
 
   Trace::ReturnTrace();
 }
 
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     const uint8_t* packet,
     size_t length,
@@ skipping 16 matching lines @@
 
   return rtc::Optional<RtpPacketReceived>(std::move(parsed_packet));
 }
 
 void Call::UpdateHistograms() {
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.LifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 }
 
-void Call::UpdateSendHistograms() {
-  if (first_packet_sent_ms_ == -1)
+void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
+  if (first_sent_packet_ms == -1)
     return;
   int64_t elapsed_sec =
-      (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
+      (clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats send_bitrate_stats =
       estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
   if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
                                 send_bitrate_stats.average);
     LOG(LS_INFO) << "WebRTC.Call.EstimatedSendBitrateInKbps, "
                  << send_bitrate_stats.ToString();
@@ skipping 559 matching lines @@
     aggregate_state = kNetworkUp;
   }
 
   LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
                << (aggregate_state == kNetworkUp ? "up" : "down");
 
   transport_send_->send_side_cc()->SignalNetworkState(aggregate_state);
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
-  if (first_packet_sent_ms_ == -1)
-    first_packet_sent_ms_ = clock_->TimeInMilliseconds();
   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
                                         clock_->TimeInMilliseconds());
   transport_send_->send_side_cc()->OnSentPacket(sent_packet);
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt_ms,
                             int64_t probing_interval_ms) {
   // TODO(perkj): Consider making sure CongestionController operates on
@@ skipping 256 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc