mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 02:30:35 +00:00
webrtcbin, rtpbin: check before setting properties on jitterbuffer
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:
parent
c726add352
commit
91317aacaf
2 changed files with 19 additions and 8 deletions
|
@ -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);
|
||||
|
||||
|
|
|
@ -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"))
|
||||
|
|
Loading…
Reference in a new issue