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}
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WebRTC LUCI CQ
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@ -287,6 +287,8 @@ void AcmReceiver::GetNetworkStatistics(
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acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
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acm_stat->jitterBufferTargetDelayMs =
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neteq_lifetime_stat.jitter_buffer_target_delay_ms;
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acm_stat->jitterBufferMinimumDelayMs =
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neteq_lifetime_stat.jitter_buffer_minimum_delay_ms;
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acm_stat->jitterBufferEmittedCount =
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neteq_lifetime_stat.jitter_buffer_emitted_count;
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acm_stat->delayedPacketOutageSamples =
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