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webrtcbin, rtpbin: check before setting properties on jitterbuffer

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In rtpbin we already systematically check for all property names
except latency, correct that.

In webrtcbin we need to check before trying to use the do-retransmission
property.

This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
This commit is contained in:
Mathieu Duponchelle 2024-02-13 18:57:05 +01:00 committed by GStreamer Marge Bot
parent c726add352
commit 91317aacaf
2 changed files with 19 additions and 8 deletions
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23
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -7866,6 +7866,7 @@ jitter_buffer_set_retransmission (SsrcMapItem * item,
{
GstWebRTCRTPTransceiver *trans;
gboolean do_nack;
GObjectClass *jb_class;
if (item->media_idx == -1)
return TRUE;
@ -7876,13 +7877,23 @@ jitter_buffer_set_retransmission (SsrcMapItem * item,
return TRUE;
}
jb_class = G_OBJECT_GET_CLASS (G_OBJECT (data->jitterbuffer));
do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack;
/* We don't set do-retransmission on rtpbin as we want per-session control */
GST_LOG_OBJECT (data->webrtc, "setting do-nack=%s for transceiver %"
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
" rtp session %u ssrc %u", do_nack ? "true" : "false", trans,
data->stream, data->stream->session_id, data->ssrc);
g_object_set (data->jitterbuffer, "do-retransmission", do_nack, NULL);
if (g_object_class_find_property (jb_class, "do-retransmission")) {
/* We don't set do-retransmission on rtpbin as we want per-session control */
GST_LOG_OBJECT (data->webrtc, "setting do-nack=%s for transceiver %"
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
" rtp session %u ssrc %u", do_nack ? "true" : "false", trans,
data->stream, data->stream->session_id, data->ssrc);
g_object_set (data->jitterbuffer, "do-retransmission", do_nack, NULL);
} else if (do_nack) {
GST_WARNING_OBJECT (data->webrtc, "Not setting do-nack for transceiver %"
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
" rtp session %u ssrc %u"
" as its jitterbuffer does not have a do-retransmission property",
trans, data->stream, data->stream->session_id, data->ssrc);
}
g_weak_ref_set (&item->rtpjitterbuffer, data->jitterbuffer);

4
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpbin.c
View file

@ -1967,8 +1967,8 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
/* configure latency and packet lost */
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
if (g_object_class_find_property (jb_class, "latency"))
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
if (g_object_class_find_property (jb_class, "drop-on-latency"))
g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
if (g_object_class_find_property (jb_class, "do-lost"))