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webrtcbin: Sync to the clock per stream and not per bundle

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By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.

Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
This commit is contained in:
Sebastian Dröge 2021-06-24 13:17:09 +03:00 committed by GStreamer Marge Bot
parent ee0124cb36
commit 0e559fc2f3
2 changed files with 42 additions and 26 deletions
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66
ext/webrtc/gstwebrtcbin.c
View file

@ -4045,37 +4045,39 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
/*
* Not-bundle case:
*
* ,-------------------------webrtcbin-------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ; ; ; ; ;
* ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
* ; sink_%u ; ; '---------------------' ;
* o----------o send_rtp_sink_%u ; ;
* ; '--------------------' ;
* '--------------------- -------------------------------------'
* ,--------------------------------------------webrtcbin-------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ,---clocksync---, ; ; ; ; ;
* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
* ; sink_%u ; ; ; ; '---------------------' ;
* o---------o sink src o---o send_rtp_sink_%u ; ;
* ; '---------------' '--------------------' ;
* '------------------------------------------------------------------------------'
*/
/*
* Bundle case:
* ,--------------------------------webrtcbin--------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ; ; ; ; ;
* ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
* ; sink_%u ,---funnel---, ; ; '---------------------' ;
* o---------o sink_%u ; ; ; ;
* ; sink_%u ; src o-o send_rtp_sink_%u ; ;
* o---------o sink_%u ; ; ; ;
* ; '------------' '--------------------' ;
* '-------------------------------------------------------------------------'
* ,-----------------------------------------------------webrtcbin--------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ; ; ; ; ;
* ; sink_%u ,---clocksync---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
* o----------o sink src o---o sink_%u ; ; ; '---------------------' ;
* ; '---------------' ; ; ; ; ;
* ; ; src o-o send_rtp_sink_%u ; ;
* ; sink_%u ,---clocksync---, ; ; ; ; ;
* o----------o sink src o---o sink%u ; '--------------------' ;
* ; '---------------' '------------' ;
* '----------------------------------------------------------------------------------------------'
*/
GstPadTemplate *rtp_templ;
GstPad *rtp_sink;
GstPad *rtp_sink, *sinkpad, *srcpad;
gchar *pad_name;
WebRTCTransceiver *trans;
GstElement *clocksync;
g_return_val_if_fail (pad->trans != NULL, NULL);
@ -4085,6 +4087,14 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
g_assert (trans->stream);
clocksync = gst_element_factory_make ("clocksync", NULL);
g_object_set (clocksync, "sync", TRUE, NULL);
gst_bin_add (GST_BIN (webrtc), clocksync);
gst_element_sync_state_with_parent (clocksync);
srcpad = gst_element_get_static_pad (clocksync, "src");
sinkpad = gst_element_get_static_pad (clocksync, "sink");
if (!webrtc->rtpfunnel) {
rtp_templ =
_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
@ -4095,9 +4105,11 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
rtp_sink =
gst_element_request_pad (webrtc->rtpbin, rtp_templ, pad_name, NULL);
g_free (pad_name);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), rtp_sink);
gst_pad_link (srcpad, rtp_sink);
gst_object_unref (rtp_sink);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), sinkpad);
pad_name = g_strdup_printf ("send_rtp_src_%u", pad->trans->mline);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (trans->stream->send_bin), "rtp_sink"))
@ -4108,12 +4120,16 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
GstPad *funnel_sinkpad =
gst_element_request_pad_simple (webrtc->rtpfunnel, pad_name);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), funnel_sinkpad);
gst_pad_link (srcpad, funnel_sinkpad);
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), sinkpad);
g_free (pad_name);
gst_object_unref (funnel_sinkpad);
}
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
gst_element_sync_state_with_parent (GST_ELEMENT (trans->stream->send_bin));
return GST_PAD (pad);

2
gst-libs/gst/webrtc/dtlstransport.c
View file

@ -184,7 +184,7 @@ gst_webrtc_dtls_transport_constructed (GObject * object)
webrtc->dtlssrtpenc = gst_element_factory_make ("dtlssrtpenc", NULL);
g_object_set (webrtc->dtlssrtpenc, "connection-id", connection_id,
"is-client", webrtc->client, "rtp-sync", TRUE, NULL);
"is-client", webrtc->client, "rtp-sync", FALSE, NULL);
webrtc->dtlssrtpdec = gst_element_factory_make ("dtlssrtpdec", NULL);
g_object_set (webrtc->dtlssrtpdec, "connection-id", connection_id, NULL);