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[Stats] Add googTimingFrameInfo to the modern API.

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This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.

Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}
This commit is contained in:
Henrik Boström
2022-10-19 17:50:09 +02:00
committed by WebRTC LUCI CQ
parent 8e7a105c51
commit c5f8f800a2
5 changed files with 54 additions and 1 deletions
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7
api/stats/rtcstats_objects.h
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@ -502,6 +502,13 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
RTCStatsMember<uint32_t> pli_count;
RTCStatsMember<uint32_t> nack_count;
RTCStatsMember<uint64_t> qp_sum;
// This is a remnant of the legacy getStats() API. When the "video-timing"
// header extension is used,
// https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/,
// `googTimingFrameInfo` is exposed with the value of
// TimingFrameInfo::ToString().
// TODO(https://crbug.com/webrtc/14586): Unship or standardize this metric.
RTCStatsMember<std::string> goog_timing_frame_info;
// Non-standard audio metrics.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;