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Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API

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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}
This commit is contained in:
Artem Titov
2020-03-11 11:18:54 +01:00
committed by Commit Bot
parent c46385c346
commit e618cc9c1e
17 changed files with 59 additions and 7 deletions
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8
api/stats/rtcstats_objects.h
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@ -327,6 +327,14 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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;