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}

modules/audio_coding/neteq/decision_logic.cc

@@ -227,6 +227,10 @@ int DecisionLogic::TargetLevelMs() const {
   return target_delay_ms;
 }
 
+int DecisionLogic::UnlimitedTargetLevelMs() const {
+  return delay_manager_->UnlimitedTargetLevelMs();
+}
+
 int DecisionLogic::GetFilteredBufferLevel() const {
   if (config_.enable_stable_playout_delay) {
     return last_playout_delay_ms_ * sample_rate_khz_;