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

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/3994>

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
index 0491733..db922f0 100644 (file)
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
@@ -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.s = s;
+  ts_stats.clock_rate = clock_rate;
 
   transport_stream_find_ssrc_map_item (ts_stats.stream, &ts_stats,
       (FindSsrcMapFunc) webrtc_stats_get_from_transport);