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webrtcbin: Remove queue after rtpfunnel

Browse source 
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.

Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
This commit is contained in:
Jan Schmidt 2022-11-19 19:22:17 +11:00 committed by GStreamer Marge Bot
parent 8177588250
commit dfb5e3365e
1 changed files with 5 additions and 11 deletions
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16
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
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@ -5782,10 +5782,9 @@ static void
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id) _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
{ {
gchar *pad_name; gchar *pad_name;
GstPad *queue_srcpad; GstPad *srcpad;
GstPad *rtp_sink; GstPad *rtp_sink;
TransportStream *stream = _find_transport_for_session (webrtc, session_id); TransportStream *stream = _find_transport_for_session (webrtc, session_id);
GstElement *queue;
g_assert (stream); g_assert (stream);
@ -5796,19 +5795,14 @@ _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
gst_bin_add (GST_BIN (webrtc), webrtc->rtpfunnel); gst_bin_add (GST_BIN (webrtc), webrtc->rtpfunnel);
gst_element_sync_state_with_parent (webrtc->rtpfunnel); gst_element_sync_state_with_parent (webrtc->rtpfunnel);
queue = gst_element_factory_make ("queue", NULL); srcpad = gst_element_get_static_pad (webrtc->rtpfunnel, "src");
gst_bin_add (GST_BIN (webrtc), queue);
gst_element_sync_state_with_parent (queue);
gst_element_link (webrtc->rtpfunnel, queue);
queue_srcpad = gst_element_get_static_pad (queue, "src");
pad_name = g_strdup_printf ("send_rtp_sink_%d", session_id); pad_name = g_strdup_printf ("send_rtp_sink_%d", session_id);
rtp_sink = gst_element_request_pad_simple (webrtc->rtpbin, pad_name); rtp_sink = gst_element_request_pad_simple (webrtc->rtpbin, pad_name);
g_free (pad_name); g_free (pad_name);
gst_pad_link (queue_srcpad, rtp_sink);
gst_object_unref (queue_srcpad); gst_pad_link (srcpad, rtp_sink);
gst_object_unref (srcpad);
gst_object_unref (rtp_sink); gst_object_unref (rtp_sink);
pad_name = g_strdup_printf ("send_rtp_src_%d", session_id); pad_name = g_strdup_printf ("send_rtp_src_%d", session_id);