commit 382f21cd9c4e41229bc0f0a1e65017cbadd343e9
author saza <saza@webrtc.org> Tue Jul 04 08:11:49 2017
committer Commit Bot <commit-bot@chromium.org> Tue Jul 04 08:11:49 2017
tree 29d0ae808012cc480cf4d35bb8016f21f3ffb353
parent 746749237ab5e34bd6bfa9cc0da63fffce528901

Revert of Add received audio and video call duration metrics based on packets. (patchset #4 id:140001 of https://codereview.webrtc.org/2957073002/ )

Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession

A Windows build fails with a mysterious compile error.

Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> 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
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: https://chromium.googlesource.com/external/webrtc/+/746749237ab5e34bd6bfa9cc0da63fffce528901

TBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882

Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}

webrtc/call/call.cc

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 1caf0d2..5c6f427 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -325,10 +325,6 @@
   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.
@@ -534,16 +530,6 @@
 }
 
 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();
@@ -1331,11 +1317,6 @@
       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) {
@@ -1343,11 +1324,6 @@
       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 f876621..f51f483 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2644,8 +2644,6 @@
 
   // 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"));