Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
Bug: webrtc:11381
Change-Id: I7df3450e50da49d178e1e3a5d9f4970672d91aac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30758}
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 197a810..e83c83d 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -327,6 +327,14 @@
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
+ // Non-standard metric showing target delay of jitter buffer.
+ // This value is increased by the target jitter buffer delay every time a
+ // sample is emitted by the jitter buffer. The added target is the target
+ // delay, in seconds, at the time that the sample was emitted from the jitter
+ // buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
+ // Currently it is implemented only for audio.
+ // TODO(titovartem) implement for video streams when will be requested.
+ RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/henbos/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;