From: Sebastian Dröge Date: Thu, 24 Jun 2021 10:17:09 +0000 (+0300) Subject: webrtcbin: Sync to the clock per stream and not per bundle X-Git-Tag: 1.19.3~507^2~262 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=0e559fc2f394114d4ef884f375f470125e28bc89;p=platform%2Fupstream%2Fgstreamer.git webrtcbin: Sync to the clock per stream and not per bundle 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: --- diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index 8af9707..88f956b 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -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); diff --git a/gst-libs/gst/webrtc/dtlstransport.c b/gst-libs/gst/webrtc/dtlstransport.c index e127555..c401f7f 100644 --- a/gst-libs/gst/webrtc/dtlstransport.c +++ b/gst-libs/gst/webrtc/dtlstransport.c @@ -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);