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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index b4a9456d77546c02b29d6672d8873409680fc2f8..f0903cf749d2f3168ffc5254dc1ff4d05021e635 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -340,6 +340,9 @@ class Call : public webrtc::Call,
   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
@@ -530,6 +533,11 @@ void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
 }
 
 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();
@@ -1317,6 +1325,9 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
       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) {