Expose relative packet arrival delay metric in stats API.
The metric is non-standard and documented in: https://github.com/henbos/webrtc-provisional-stats/pull/14
Bug: webrtc:10333
Change-Id: Ie5b4bbad5b1e2c9104742931529bab8f48f51f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26999}diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 8c1ab02..64a5ec6 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -1428,6 +1428,7 @@
voice_receiver_info.jitter_buffer_emitted_count = 13;
voice_receiver_info.jitter_buffer_flushes = 7;
voice_receiver_info.delayed_packet_outage_samples = 15;
+ voice_receiver_info.relative_packet_arrival_delay_seconds = 16;
stats_->CreateMockRtpSendersReceiversAndChannels(
{}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
@@ -1464,6 +1465,7 @@
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
expected_remote_audio_track.jitter_buffer_flushes = 7;
expected_remote_audio_track.delayed_packet_outage_samples = 15;
+ expected_remote_audio_track.relative_packet_arrival_delay = 16;
ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
EXPECT_EQ(expected_remote_audio_track,
report->Get(expected_remote_audio_track.id())