Add a histogram metric tracking for how long audio RTP packets are sent

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 0bddde9fc612739cc741b122029ea4bd54c87685..ac3c146c4071e57c99218a3c176a683bd9aa4935 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -46,6 +46,7 @@
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/rtc_base/thread_checker.h"
+#include "webrtc/rtc_base/time_interval.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
@@ -329,6 +330,7 @@ class Call : public webrtc::Call,
   rtc::Optional<int64_t> last_received_rtp_audio_ms_;
   rtc::Optional<int64_t> first_received_rtp_video_ms_;
   rtc::Optional<int64_t> last_received_rtp_video_ms_;
+  rtc::TimeInterval sent_rtp_audio_timer_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
@@ -510,6 +512,12 @@ void Call::UpdateHistograms() {
 void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
   if (first_sent_packet_ms == -1)
     return;
+  if (!sent_rtp_audio_timer_ms_.Empty()) {
+    RTC_HISTOGRAM_COUNTS_100000(

[pbos-webrtc, 2017/07/13 16:59:45]

Is this code actually intending to track sent RTP from multiple send streams?
Otherwise this could be done entirely in AudioSendStream?

Would it be better for the transport object to track for how long each payload
type is sent, construct the lifetime through there and log lifetime for both
audio or video? E.g. on destruction, report for how long each payload type is
sent.

Also, what should happen when a stream is inactive for 18/20 minutes in the
middle? Is that something we care about? This stats will report that it's
sending for 20 minutes.

+        "WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds",
+        (*sent_rtp_audio_timer_ms_.Last() - *sent_rtp_audio_timer_ms_.First()) /
+            1000);
+  }
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
@@ -648,6 +656,12 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
     }
   }
   UpdateAggregateNetworkState();
+  const rtc::TimeInterval* stream_active_lifetime =
+      audio_send_stream->GetActiveLifetime();
+  if (!stream_active_lifetime->Empty()) {
+    sent_rtp_audio_timer_ms_.Extend(*stream_active_lifetime->First());
+    sent_rtp_audio_timer_ms_.Extend(*stream_active_lifetime->Last());
+  }
   delete audio_send_stream;
 }