commit 0d7f04daa0ddc621a7e3d6a0373b3ae458511030
author saza <saza@webrtc.org> Tue Jul 04 11:05:06 2017
committer Commit Bot <commit-bot@chromium.org> Tue Jul 04 11:05:06 2017
tree 02e07bb0994d16f17346188a35bf5ad2e0546213
parent 38fecafa48ff6abfdf0279ce3c04df7e26401e08

Reland of Add received audio/video call duration metrics based on packets.

Original issue:
https://codereview.webrtc.org/2957073002/

Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.

Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.

BUG=webrtc:7882
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}

webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5c6f427..1caf0d2 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -325,6 +325,10 @@
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
+  rtc::Optional<int64_t> first_received_rtp_audio_ms_;
+  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_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
@@ -530,6 +534,16 @@
 }
 
 void Call::UpdateReceiveHistograms() {
+  if (first_received_rtp_audio_ms_) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
+        (*last_received_rtp_audio_ms_ - *first_received_rtp_audio_ms_) / 1000);
+  }
+  if (first_received_rtp_video_ms_) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
+        (*last_received_rtp_video_ms_ - *first_received_rtp_video_ms_) / 1000);
+  }
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats video_bytes_per_sec =
       received_video_bytes_per_second_counter_.GetStats();
@@ -1317,6 +1331,11 @@
       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);
+      const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
+      if (!first_received_rtp_audio_ms_) {
+        first_received_rtp_audio_ms_.emplace(arrival_time_ms);
+      }
+      last_received_rtp_audio_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   } else if (media_type == MediaType::VIDEO) {
@@ -1324,6 +1343,11 @@
       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);
+      const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
+      if (!first_received_rtp_video_ms_) {
+        first_received_rtp_video_ms_.emplace(arrival_time_ms);
+      }
+      last_received_rtp_video_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   }

webrtc/video/end_to_end_tests.cc

diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index f51f483..f876621 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2644,6 +2644,8 @@
 
   // Verify that stats have been updated once.
   EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
+  EXPECT_EQ(1, metrics::NumSamples(
+                   "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));