mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-30 07:42:32 +00:00
webrtcbin: fix ulpfec / red for the BUNDLE case
* Add fec / red encoders as direct children of webrtcbin, instead of providing them to rtpbin through the request-fec-encoder signal. That is because they need to be placed before the rtpfunnel, which is placed upstream of rtpbin. * Update configuration of red decoders to set a list of RED payloads on them, instead of setting the pt property. That is because there may be one RED pt per media in the same session. * Connect to request-fec-decoder-full instead of request-fec-decoder, in order to instantiate FEC decoders according to the payload type of the stream. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
This commit is contained in:
Mathieu Duponchelle
GStreamer Marge Bot
parent
d12d45db77
commit
06893b8b5e
3 changed files with 176 additions and 128 deletions
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@ -4342,45 +4342,123 @@ out:
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return ret;
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}
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static GstElement *
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_build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans)
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{
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GstElement *ret = NULL;
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GstElement *prev = NULL;
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guint ulpfec_pt = 0;
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guint red_pt = 0;
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GstPad *sinkpad = NULL;
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GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
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if (trans->stream) {
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ulpfec_pt =
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transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline);
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red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline);
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}
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if (ulpfec_pt || red_pt)
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ret = gst_bin_new (NULL);
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if (ulpfec_pt) {
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GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
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GstCaps *caps = transport_stream_get_caps_for_pt (trans->stream, ulpfec_pt);
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GST_DEBUG_OBJECT (webrtc,
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"Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline,
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ulpfec_pt);
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gst_bin_add (GST_BIN (ret), fecenc);
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sinkpad = gst_element_get_static_pad (fecenc, "sink");
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g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
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trans->fec_percentage, NULL);
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if (caps && !gst_caps_is_empty (caps)) {
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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const gchar *media = gst_structure_get_string (s, "media");
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if (!g_strcmp0 (media, "video"))
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g_object_set (fecenc, "multipacket", TRUE, NULL);
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}
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prev = fecenc;
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}
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if (red_pt) {
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GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
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GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d",
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rtp_trans->mline, red_pt);
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gst_bin_add (GST_BIN (ret), redenc);
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if (prev)
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gst_element_link (prev, redenc);
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else
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sinkpad = gst_element_get_static_pad (redenc, "sink");
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g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
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prev = redenc;
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}
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if (sinkpad) {
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GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
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gst_object_unref (sinkpad);
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gst_element_add_pad (ret, ghost);
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}
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if (prev) {
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GstPad *srcpad = gst_element_get_static_pad (prev, "src");
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GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
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gst_object_unref (srcpad);
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gst_element_add_pad (ret, ghost);
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}
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return ret;
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}
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static GstPad *
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_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
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{
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/*
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* Not-bundle case:
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*
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* ,--------------------------------------------webrtcbin-------------------------,
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* ; ;
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* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
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* ; ; send_rtp_src_%u o---o rtp_sink ; ;
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* ; ,---clocksync---, ; ; ; ; ;
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* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
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* ; sink_%u ; ; ; ; '---------------------' ;
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* o---------o sink src o---o send_rtp_sink_%u ; ;
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* ; '---------------' '--------------------' ;
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* '------------------------------------------------------------------------------'
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* ,--------------------------------------------webrtcbin--------------------------------------------,
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* ; ;
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* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
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* ; ; send_rtp_src_%u o---o rtp_sink ; ;
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* ; ,---clocksync---, ; ; ; ; ;
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* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
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* ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ;
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* o---------o sink src o-o sink src o--o send_rtp_sink_%u ; ;
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* ; '---------------' ,-----------------, '--------------------' ;
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* '-------------------------------------------------------------------------------------------------'
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*/
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/*
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* Bundle case:
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* ,-----------------------------------------------------webrtcbin--------------------------------,
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* ; ;
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* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
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* ; ; send_rtp_src_%u o---o rtp_sink ; ;
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* ; ; ; ; ; ;
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* ; sink_%u ,---clocksync---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
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* o----------o sink src o---o sink_%u ; ; ; '---------------------' ;
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* ; '---------------' ; ; ; ; ;
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* ; ; src o-o send_rtp_sink_%u ; ;
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* ; sink_%u ,---clocksync---, ; ; ; ; ;
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* o----------o sink src o---o sink%u ; '--------------------' ;
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* ; '---------------' '------------' ;
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* '----------------------------------------------------------------------------------------------'
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* ,-----------------------------------------------------webrtcbin---------------------------------------------------,
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* ; ;
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* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
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* ; ; send_rtp_src_%u o---o rtp_sink ; ;
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* ; ; ; ; ; ;
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* ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
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* o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ;
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* ; '---------------' ,-----------------, ; ; ; ; ;
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* ; ; src o-o send_rtp_sink_%u ; ;
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* ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ;
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* o----------o sink src o-o sink src o--o sink%u ; '--------------------' ;
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* ; '---------------' ,-----------------, '------------' ;
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* '-----------------------------------------------------------------------------------------------------------------'
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*/
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GstPadTemplate *rtp_templ;
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GstPad *rtp_sink, *sinkpad, *srcpad;
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gchar *pad_name;
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WebRTCTransceiver *trans;
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GstElement *clocksync;
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GstElement *fec_encoder;
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g_return_val_if_fail (pad->trans != NULL, NULL);
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@ -4398,6 +4476,19 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
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srcpad = gst_element_get_static_pad (clocksync, "src");
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sinkpad = gst_element_get_static_pad (clocksync, "sink");
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if ((fec_encoder = _build_fec_encoder (webrtc, trans))) {
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GstPad *fec_sink;
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gst_bin_add (GST_BIN (webrtc), fec_encoder);
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gst_element_sync_state_with_parent (fec_encoder);
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fec_sink = gst_element_get_static_pad (fec_encoder, "sink");
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gst_pad_link (srcpad, fec_sink);
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gst_object_unref (srcpad);
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gst_object_unref (fec_sink);
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srcpad = gst_element_get_static_pad (fec_encoder, "src");
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}
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if (!webrtc->rtpfunnel) {
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rtp_templ =
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_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
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@ -4681,6 +4772,7 @@ _update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
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}
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item.pt = pt;
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item.media_idx = media_idx;
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gst_caps_unref (outcaps);
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g_array_append_val (stream->ptmap, item);
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@ -6359,7 +6451,7 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
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stream = _find_transport_for_session (webrtc, session_id);
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if (stream)
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have_rtx = transport_stream_get_pt (stream, "RTX") != 0;
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have_rtx = transport_stream_get_pt (stream, "RTX", -1) != 0;
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GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
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" with pt map %" GST_PTR_FORMAT, stream, pt_map);
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@ -6428,20 +6520,43 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
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GstElement *prev = NULL;
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GstPad *sinkpad = NULL;
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TransportStream *stream;
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gint red_pt = 0;
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gint rtx_pt = 0;
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GValue red_pt_array = { 0, };
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gboolean have_red_pt = FALSE;
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g_value_init (&red_pt_array, GST_TYPE_ARRAY);
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stream = _find_transport_for_session (webrtc, session_id);
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if (stream) {
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red_pt = transport_stream_get_pt (stream, "RED");
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rtx_pt = transport_stream_get_pt (stream, "RTX");
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guint i = 0;
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for (i = 0; i < stream->ptmap->len; i++) {
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PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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if (!gst_caps_is_empty (item->caps)) {
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GstStructure *s = gst_caps_get_structure (item->caps, 0);
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if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RED")) {
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GValue ptval = { 0, };
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g_value_init (&ptval, G_TYPE_INT);
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g_value_set_int (&ptval, item->pt);
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gst_value_array_append_value (&red_pt_array, &ptval);
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g_value_unset (&ptval);
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have_red_pt = TRUE;
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}
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}
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}
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rtx_pt = transport_stream_get_pt (stream, "RTX", -1);
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}
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GST_LOG_OBJECT (webrtc, "requesting aux receiver for stream %" GST_PTR_FORMAT,
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stream);
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if (red_pt || rtx_pt)
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if (have_red_pt || rtx_pt)
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ret = gst_bin_new (NULL);
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if (rtx_pt) {
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@ -6461,15 +6576,14 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
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prev = gst_object_ref (stream->rtxreceive);
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}
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if (red_pt) {
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if (have_red_pt) {
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GstElement *rtpreddec = gst_element_factory_make ("rtpreddec", NULL);
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GST_DEBUG_OBJECT (webrtc, "Creating RED decoder for pt %d in session %u",
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red_pt, session_id);
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GST_DEBUG_OBJECT (webrtc, "Creating RED decoder in session %u", session_id);
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gst_bin_add (GST_BIN (ret), rtpreddec);
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g_object_set (rtpreddec, "pt", red_pt, NULL);
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g_object_set_property (G_OBJECT (rtpreddec), "payloads", &red_pt_array);
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if (prev)
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gst_element_link (prev, rtpreddec);
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@ -6497,6 +6611,7 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
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}
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out:
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g_value_unset (&red_pt_array);
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return ret;
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error:
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@ -6506,12 +6621,12 @@ error:
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}
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static GstElement *
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on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
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GstWebRTCBin * webrtc)
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on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id,
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guint ssrc, guint pt, GstWebRTCBin * webrtc)
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{
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TransportStream *stream;
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GstElement *ret = NULL;
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gint pt = 0;
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gint fec_pt = 0;
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GObject *internal_storage;
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stream = _find_transport_for_session (webrtc, session_id);
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@ -6526,105 +6641,33 @@ on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
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* we detect it (by connecting to ptdemux:new-payload-type for
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* example)
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*/
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if (stream)
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pt = transport_stream_get_pt (stream, "ULPFEC");
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if (stream) {
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guint i;
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if (pt) {
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for (i = 0; i < stream->ptmap->len; i++) {
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PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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if (item->pt == pt) {
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fec_pt = transport_stream_get_pt (stream, "ULPFEC", item->media_idx);
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break;
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}
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}
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}
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if (fec_pt) {
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GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
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pt, session_id);
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fec_pt, session_id);
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ret = gst_element_factory_make ("rtpulpfecdec", NULL);
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g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
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&internal_storage);
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g_object_set (ret, "pt", pt, "storage", internal_storage, NULL);
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g_object_set (ret, "pt", fec_pt, "storage", internal_storage, NULL);
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g_object_unref (internal_storage);
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}
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return ret;
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}
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static GstElement *
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on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
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GstWebRTCBin * webrtc)
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{
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GstElement *ret = NULL;
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GstElement *prev = NULL;
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TransportStream *stream;
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guint ulpfec_pt = 0;
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guint red_pt = 0;
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GstPad *sinkpad = NULL;
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GstWebRTCRTPTransceiver *trans;
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stream = _find_transport_for_session (webrtc, session_id);
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trans = _find_transceiver (webrtc, &session_id,
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(FindTransceiverFunc) transceiver_match_for_mline);
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if (stream) {
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ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC");
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red_pt = transport_stream_get_pt (stream, "RED");
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}
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if (ulpfec_pt || red_pt)
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ret = gst_bin_new (NULL);
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if (ulpfec_pt) {
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GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
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GstCaps *caps = transport_stream_get_caps_for_pt (stream, ulpfec_pt);
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GST_DEBUG_OBJECT (webrtc,
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"Creating ULPFEC encoder for session %d with pt %d", session_id,
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ulpfec_pt);
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gst_bin_add (GST_BIN (ret), fecenc);
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sinkpad = gst_element_get_static_pad (fecenc, "sink");
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g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
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WEBRTC_TRANSCEIVER (trans)->fec_percentage, NULL);
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if (caps && !gst_caps_is_empty (caps)) {
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const GstStructure *s = gst_caps_get_structure (caps, 0);
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const gchar *media = gst_structure_get_string (s, "media");
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if (!g_strcmp0 (media, "video"))
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g_object_set (fecenc, "multipacket", TRUE, NULL);
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}
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prev = fecenc;
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}
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if (red_pt) {
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GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
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GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for session %d with pt %d",
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session_id, red_pt);
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gst_bin_add (GST_BIN (ret), redenc);
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if (prev)
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gst_element_link (prev, redenc);
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else
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sinkpad = gst_element_get_static_pad (redenc, "sink");
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g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
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prev = redenc;
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}
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if (sinkpad) {
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GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
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gst_object_unref (sinkpad);
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gst_element_add_pad (ret, ghost);
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}
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if (prev) {
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GstPad *srcpad = gst_element_get_static_pad (prev, "src");
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GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
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gst_object_unref (srcpad);
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gst_element_add_pad (ret, ghost);
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}
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return ret;
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}
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static void
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on_rtpbin_bye_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
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GstWebRTCBin * webrtc)
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@ -6788,10 +6831,8 @@ _create_rtpbin (GstWebRTCBin * webrtc)
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G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc);
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g_signal_connect (rtpbin, "new-storage",
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G_CALLBACK (on_rtpbin_new_storage), webrtc);
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g_signal_connect (rtpbin, "request-fec-decoder",
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G_CALLBACK (on_rtpbin_request_fec_decoder), webrtc);
|
||||
g_signal_connect (rtpbin, "request-fec-encoder",
|
||||
G_CALLBACK (on_rtpbin_request_fec_encoder), webrtc);
|
||||
g_signal_connect (rtpbin, "request-fec-decoder-full",
|
||||
G_CALLBACK (on_rtpbin_request_fec_decoder_full), webrtc);
|
||||
g_signal_connect (rtpbin, "on-bye-ssrc",
|
||||
G_CALLBACK (on_rtpbin_bye_ssrc), webrtc);
|
||||
g_signal_connect (rtpbin, "on-bye-timeout",
|
||||
|
|
|
|||
|
|
@ -55,13 +55,18 @@ transport_stream_get_caps_for_pt (TransportStream * stream, guint pt)
|
|||
}
|
||||
|
||||
int
|
||||
transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name)
|
||||
transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name,
|
||||
guint media_idx)
|
||||
{
|
||||
guint i;
|
||||
gint ret = 0;
|
||||
|
||||
for (i = 0; i < stream->ptmap->len; i++) {
|
||||
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
|
||||
|
||||
if (media_idx != -1 && media_idx != item->media_idx)
|
||||
continue;
|
||||
|
||||
if (!gst_caps_is_empty (item->caps)) {
|
||||
GstStructure *s = gst_caps_get_structure (item->caps, 0);
|
||||
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
|
||||
|
|
|
|||
|
|
@ -34,6 +34,7 @@ GType transport_stream_get_type(void);
|
|||
typedef struct
|
||||
{
|
||||
guint8 pt;
|
||||
guint media_idx;
|
||||
GstCaps *caps;
|
||||
} PtMapItem;
|
||||
|
||||
|
|
@ -76,7 +77,8 @@ struct _TransportStreamClass
|
|||
TransportStream * transport_stream_new (GstWebRTCBin * webrtc,
|
||||
guint session_id);
|
||||
int transport_stream_get_pt (TransportStream * stream,
|
||||
const gchar * encoding_name);
|
||||
const gchar * encoding_name,
|
||||
guint media_idx);
|
||||
int * transport_stream_get_all_pt (TransportStream * stream,
|
||||
const gchar * encoding_name,
|
||||
gsize * pt_len);
|
||||
|
|
|
|||
Loading…
Reference in a new issue