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/statistics_calculator.h

@@ -84,7 +84,8 @@ class StatisticsCalculator {
   // Update jitter buffer delay counter.
   void JitterBufferDelay(size_t num_samples,
                          uint64_t waiting_time_ms,
-                         uint64_t target_delay_ms);
+                         uint64_t target_delay_ms,
+                         uint64_t unlimited_target_delay_ms);
 
   // Stores new packet waiting time in waiting time statistics.
   void StoreWaitingTime(int waiting_time_ms);