Add received audio and video call duration metrics based on packets.

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 307 matching lines @@
 
   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.
   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_;
 

[aleloi, 2017/06/27 14:13:37]

I think the rtp_audio_ms_ counters should be moved here, because they are
accessed in the same pattern as the RateCounters. The synchronization comment
applies to them too.

   // 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_);
   AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
   AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
 
   std::map<std::string, rtc::NetworkRoute> network_routes_;
 
   std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
   ReceiveSideCongestionController receive_side_cc_;
   const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
   const int64_t start_ms_;
+  int64_t first_received_rtp_audio_ms_ = -1;
+  int64_t last_received_rtp_audio_ms_ = -1;

[aleloi, 2017/06/27 14:13:37]

There is a type rtc::Optional from webrtc/base/optional.h. Can you use that
instead of a special value to distinguish between 'no value yet' and valid
value?

+
   // TODO(perkj): |worker_queue_| is supposed to replace
   // |module_process_thread_|.
   // |worker_queue| is defined last to ensure all pending tasks are cancelled
   // and deleted before any other members.
   rtc::TaskQueue worker_queue_;
 
   // The config mask set by SetBitrateConfigMask.
   // 0 <= min <= start <= max
   Config::BitrateConfigMask bitrate_config_mask_;
 
@@ skipping 170 matching lines @@
       pacer_bitrate_kbps_counter_.ProcessAndGetStats();
   if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
                                 pacer_bitrate_stats.average);
     LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, "
                  << pacer_bitrate_stats.ToString();
   }
 }
 
 void Call::UpdateReceiveHistograms() {
+  if (first_received_rtp_audio_ms_ != -1) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
+        (last_received_rtp_audio_ms_ - first_received_rtp_audio_ms_) / 1000);
+  }
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats video_bytes_per_sec =
       received_video_bytes_per_second_counter_.GetStats();
   if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
                                 video_bytes_per_sec.average * 8 / 1000);
     LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
                  << video_bytes_per_sec.ToStringWithMultiplier(8);
   }
   AggregatedStats audio_bytes_per_sec =
@@ skipping 767 matching lines @@
   }
   parsed_packet->IdentifyExtensions(it->second.extensions);
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   if (media_type == MediaType::AUDIO) {
     if (audio_receiver_controller.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
       event_log_->LogRtpHeader(kIncomingPacket, packet, length);
+      if (first_received_rtp_audio_ms_ == -1)
+        first_received_rtp_audio_ms_ = packet_time.timestamp;

[aleloi, 2017/06/27 14:13:37]

Other places in this file use brackets for single statement if:s. Please add
brackets for consistency. (There is no style guide rule for this AFAIK).

[saza WebRTC, 2017/06/29 11:49:05]

Ok, I'll add them (for safety, if nothing else). The file is inconsistent
though, e.g. lines 1106-1116, 1196, 1258, 1353, and 1365 omit braces.

Yeah, style guide leaves it up to the project.
https://google.github.io/styleguide/cppguide.html#Conditionals

+      last_received_rtp_audio_ms_ = packet_time.timestamp;
       return DELIVERY_OK;
     }
   } else if (media_type == MediaType::VIDEO) {
     if (video_receiver_controller.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
       event_log_->LogRtpHeader(kIncomingPacket, packet, length);
       return DELIVERY_OK;
     }
   }
@@ skipping 53 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