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;
}