commit 1a84b565ac09817be1ef3f5d1f8035d85fe20a07
author Ivo Creusen <ivoc@google.com> Tue Jul 19 14:33:10 2022
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Jul 20 09:14:03 2022
tree aa7699d3ad01a4cbcc8caa2b84da6ea378b5b8c0
parent e7696f771d7391aad6b40e67fc7dc1282c22ff2c

Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

This metric was recently added to the standard (see https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay). This CL implements it for audio streams.

Bug: webrtc:14141
Change-Id: I79d918639cd12361ebbc28c2be41549e33fa7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262770
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37567}

pc/rtc_stats_collector.cc

diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index f79ad4e..4b92fb6 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -409,6 +409,10 @@
     inbound_stats->jitter_buffer_target_delay =
         *media_receiver_info.jitter_buffer_target_delay_seconds;
   }
+  if (media_receiver_info.jitter_buffer_minimum_delay_seconds) {
+    inbound_stats->jitter_buffer_minimum_delay =
+        *media_receiver_info.jitter_buffer_minimum_delay_seconds;
+  }
   inbound_stats->jitter_buffer_emitted_count =
       media_receiver_info.jitter_buffer_emitted_count;
   if (media_receiver_info.nacks_sent) {