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webrtcstats: Remove receiver side when sending

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Those are just invalid and just reflect what we sent. We'd need to parse the
RTCP XR packets from the other side to know more about those.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
This commit is contained in:
Olivier Crête 2020-10-09 20:45:10 -04:00 committed by GStreamer Merge Bot
parent ba0dfa52d2
commit a46c6e3a97
1 changed files with 3 additions and 12 deletions
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15
ext/webrtc/gstwebrtcstats.c
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@ -122,22 +122,13 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
/* To be added: kind */
/* RTCReceivedRtpStreamStats */
if (gst_structure_get_uint64 (source_stats, "packets-received", &packets))
gst_structure_set (r_in, "packets-received", G_TYPE_UINT64, packets,
NULL);
if (gst_structure_get_int (source_stats, "packets-lost", &lost))
gst_structure_set (r_in, "packets-lost", G_TYPE_INT, lost, NULL);
if (gst_structure_get_uint (source_stats, "jitter", &jitter))
gst_structure_set (r_in, "jitter", G_TYPE_DOUBLE,
CLOCK_RATE_VALUE_TO_SECONDS (jitter, clock_rate), NULL);
/* RTCReceivedRtpStreamStats:
To be added:
double fractionLost;
unsigned long long packetsReceived;
long long packetsLost;
double jitter;
unsigned long packetsDiscarded;
unsigned long packetsRepaired;
unsigned long burstPacketsLost;