[Stats] Add googTimingFrameInfo to the modern API.
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}
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 2b6c622..b370381 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -502,6 +502,13 @@
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;