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Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

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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}
This commit is contained in:
Ivo Creusen
2022-07-19 16:33:10 +02:00
committed by WebRTC LUCI CQ
parent e7696f771d
commit 1a84b565ac
21 changed files with 74 additions and 9 deletions
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4
pc/rtc_stats_collector.cc
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@ -409,6 +409,10 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
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) {