Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI
Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index e16c7e3..adca9ef 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1427,6 +1427,7 @@
voice_receiver_info.concealment_events = 12;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
voice_receiver_info.jitter_buffer_flushes = 7;
+ voice_receiver_info.delayed_packet_outage_samples = 15;
stats_->CreateMockRtpSendersReceiversAndChannels(
{}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
@@ -1461,6 +1462,7 @@
expected_remote_audio_track.concealment_events = 12;
expected_remote_audio_track.jitter_buffer_delay = 3456;
expected_remote_audio_track.jitter_buffer_flushes = 7;
+ expected_remote_audio_track.delayed_packet_outage_samples = 15;
ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
EXPECT_EQ(expected_remote_audio_track,
report->Get(expected_remote_audio_track.id())
