| commit | 1a84b565ac09817be1ef3f5d1f8035d85fe20a07 | [log] [tgz] |
|---|---|---|
| author | Ivo Creusen <ivoc@google.com> | Tue Jul 19 16:33:10 2022 +0200 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Jul 20 09:14:03 2022 +0000 |
| tree | aa7699d3ad01a4cbcc8caa2b84da6ea378b5b8c0 | |
| parent | e7696f771d7391aad6b40e67fc7dc1282c22ff2c [diff] [blame] |
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/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc index 1637ae7..6a6367d 100644 --- a/modules/audio_coding/neteq/neteq_impl.cc +++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -2012,7 +2012,8 @@ RTC_DCHECK(controller_); stats_->JitterBufferDelay(packet_duration, waiting_time_ms, - controller_->TargetLevelMs()); + controller_->TargetLevelMs(), + controller_->UnlimitedTargetLevelMs()); packet_list->push_back(std::move(*packet)); // Store packet in list. packet = absl::nullopt; // Ensure it's never used after the move.