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webrtc: Calculate the jitter for remote-inbound-rtp stats

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Populate the clock-rate in the internal stats structure, so
it can be used by the _get_stats_from_remote_rtp_source_stats()
method to calculate remote receivers' jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3900>
This commit is contained in:
Jan Schmidt 2023-02-07 04:58:04 +11:00
parent 615a019457
commit 621604aa3e
1 changed files with 1 additions and 0 deletions
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subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
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@ -977,6 +977,7 @@ _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
ts_stats.source_stats->n_values, ts_stats.stream->transport); ts_stats.source_stats->n_values, ts_stats.stream->transport);
ts_stats.s = s; ts_stats.s = s;
ts_stats.clock_rate = clock_rate;
transport_stream_find_ssrc_map_item (ts_stats.stream, &ts_stats, transport_stream_find_ssrc_map_item (ts_stats.stream, &ts_stats,
(FindSsrcMapFunc) webrtc_stats_get_from_transport); (FindSsrcMapFunc) webrtc_stats_get_from_transport);