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webrtcstats: RTCP computed RTT is only available at sender

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The receiver doesn't have the information to compute it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1766>
This commit is contained in:
Olivier Crête 2020-10-09 19:38:15 -04:00 committed by GStreamer Merge Bot
parent a5c3331197
commit 895ea210c2
1 changed files with 14 additions and 14 deletions
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28
ext/webrtc/gstwebrtcstats.c
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@ -105,10 +105,14 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
if (internal) { if (internal) {
GstStructure *r_in, *out; GstStructure *r_in, *out;
gchar *out_id, *r_in_id; gchar *out_id, *r_in_id;
gboolean have_rb = FALSE;
out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc); out_id = g_strdup_printf ("rtp-outbound-stream-stats_%u", ssrc);
r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc); r_in_id = g_strdup_printf ("rtp-remote-inbound-stream-stats_%u", ssrc);
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb,
NULL);
r_in = gst_structure_new_empty (r_in_id); r_in = gst_structure_new_empty (r_in_id);
_set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id); _set_base_stats (r_in, GST_WEBRTC_STATS_REMOTE_INBOUND_RTP, ts, r_in_id);
@ -142,6 +146,14 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
double gapLossRate; double gapLossRate;
double gapDiscardRate; double gapDiscardRate;
*/ */
if (have_rb) {
guint32 rtt;
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val = FIXED_16_16_TO_DOUBLE (rtt);
gst_structure_set (r_in, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
}
}
/* RTCRemoteInboundRTPStreamStats */ /* RTCRemoteInboundRTPStreamStats */
/* XXX: framesDecoded, lastPacketReceivedTimestamp */ /* XXX: framesDecoded, lastPacketReceivedTimestamp */
@ -191,10 +203,9 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
} else { } else {
GstStructure *in, *r_out; GstStructure *in, *r_out;
gchar *r_out_id, *in_id; gchar *r_out_id, *in_id;
gboolean have_rb = FALSE, have_sr = FALSE; gboolean have_sr = FALSE;
gst_structure_get (source_stats, "have-rb", G_TYPE_BOOLEAN, &have_rb, gst_structure_get (source_stats, "have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
"have-sr", G_TYPE_BOOLEAN, &have_sr, NULL);
in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc); in_id = g_strdup_printf ("rtp-inbound-stream-stats_%u", ssrc);
r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc); r_out_id = g_strdup_printf ("rtp-remote-outbound-stream-stats_%u", ssrc);
@ -251,17 +262,6 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL); gst_structure_set (r_out, "codec-id", G_TYPE_STRING, codec_id, NULL);
gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id, gst_structure_set (r_out, "transport-id", G_TYPE_STRING, transport_id,
NULL); NULL);
if (have_rb) {
guint32 rtt;
if (gst_structure_get_uint (source_stats, "rb-round-trip", &rtt)) {
/* 16.16 fixed point to double */
double val = FIXED_16_16_TO_DOUBLE (rtt);
gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, val, NULL);
}
} else {
/* default values */
gst_structure_set (r_out, "round-trip-time", G_TYPE_DOUBLE, 0.0, NULL);
}
/* XXX: mediaType, trackId, sliCount, qpSum */ /* XXX: mediaType, trackId, sliCount, qpSum */
/* RTCSentRTPStreamStats */ /* RTCSentRTPStreamStats */