From 382f21cd9c4e41229bc0f0a1e65017cbadd343e9 Mon Sep 17 00:00:00 2001 From: saza Date: Tue, 4 Jul 2017 01:11:49 -0700 Subject: [PATCH] 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 | 24 ------------------------ webrtc/video/end_to_end_tests.cc | 2 -- 2 files changed, 26 deletions(-) diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 1caf0d2376..5c6f427e3b 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -325,10 +325,6 @@ class Call : public webrtc::Call, RateCounter received_audio_bytes_per_second_counter_; RateCounter received_video_bytes_per_second_counter_; RateCounter received_rtcp_bytes_per_second_counter_; - rtc::Optional first_received_rtp_audio_ms_; - rtc::Optional last_received_rtp_audio_ms_; - rtc::Optional first_received_rtp_video_ms_; - rtc::Optional 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::UpdateSendHistograms(int64_t first_sent_packet_ms) { } 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 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, received_bytes_per_second_counter_.Add(static_cast(length)); received_audio_bytes_per_second_counter_.Add(static_cast(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 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, received_bytes_per_second_counter_.Add(static_cast(length)); received_video_bytes_per_second_counter_.Add(static_cast(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; } } diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index f876621fd3..f51f483f05 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -2644,8 +2644,6 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, // 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"));