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}
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index cd5d971..9c91c85 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -261,12 +261,16 @@
   lifetime_stats_.total_samples_received += num_samples;
 }
 
-void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
-                                             uint64_t waiting_time_ms,
-                                             uint64_t target_delay_ms) {
+void StatisticsCalculator::JitterBufferDelay(
+    size_t num_samples,
+    uint64_t waiting_time_ms,
+    uint64_t target_delay_ms,
+    uint64_t unlimited_target_delay_ms) {
   lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
   lifetime_stats_.jitter_buffer_target_delay_ms +=
       target_delay_ms * num_samples;
+  lifetime_stats_.jitter_buffer_minimum_delay_ms +=
+      unlimited_target_delay_ms * num_samples;
   lifetime_stats_.jitter_buffer_emitted_count += num_samples;
 }