From 0e559fc2f394114d4ef884f375f470125e28bc89 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Thu, 24 Jun 2021 13:17:09 +0300 Subject: 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: --- ext/webrtc/gstwebrtcbin.c | 66 +++++++++++++++++++++++++++++------------------ 1 file changed, 41 insertions(+), 25 deletions(-) (limited to 'ext') diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index 8af9707f6..88f956bcb 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); -- cgit v1.2.1