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webrtcbin: fix ulpfec / red for the BUNDLE case

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* 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 2021-12-07 23:55:22 +01:00 committed by GStreamer Marge Bot
parent d12d45db77
commit 06893b8b5e
3 changed files with 176 additions and 128 deletions
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293
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -4342,45 +4342,123 @@ out:
return ret;
}
static GstElement *
_build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans)
{
GstElement *ret = NULL;
GstElement *prev = NULL;
guint ulpfec_pt = 0;
guint red_pt = 0;
GstPad *sinkpad = NULL;
GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
if (trans->stream) {
ulpfec_pt =
transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline);
red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline);
}
if (ulpfec_pt || red_pt)
ret = gst_bin_new (NULL);
if (ulpfec_pt) {
GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
GstCaps *caps = transport_stream_get_caps_for_pt (trans->stream, ulpfec_pt);
GST_DEBUG_OBJECT (webrtc,
"Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline,
ulpfec_pt);
gst_bin_add (GST_BIN (ret), fecenc);
sinkpad = gst_element_get_static_pad (fecenc, "sink");
g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
trans->fec_percentage, NULL);
if (caps && !gst_caps_is_empty (caps)) {
const GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *media = gst_structure_get_string (s, "media");
if (!g_strcmp0 (media, "video"))
g_object_set (fecenc, "multipacket", TRUE, NULL);
}
prev = fecenc;
}
if (red_pt) {
GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d",
rtp_trans->mline, red_pt);
gst_bin_add (GST_BIN (ret), redenc);
if (prev)
gst_element_link (prev, redenc);
else
sinkpad = gst_element_get_static_pad (redenc, "sink");
g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
prev = redenc;
}
if (sinkpad) {
GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
gst_object_unref (sinkpad);
gst_element_add_pad (ret, ghost);
}
if (prev) {
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
gst_object_unref (srcpad);
gst_element_add_pad (ret, ghost);
}
return ret;
}
static GstPad *
_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
/*
* Not-bundle case:
*
* ,--------------------------------------------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 ; ;
* ; '---------------' '--------------------' ;
* '------------------------------------------------------------------------------'
* ,--------------------------------------------webrtcbin--------------------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ,---clocksync---, ; ; ; ; ;
* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
* ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ;
* o---------o sink src 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 ; ;
* ; ; ; ; ; ;
* ; 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 ; '--------------------' ;
* ; '---------------' '------------' ;
* '----------------------------------------------------------------------------------------------'
* ,-----------------------------------------------------webrtcbin---------------------------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ; ; ; ; ;
* ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
* o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ;
* ; '---------------' ,-----------------, ; ; ; ; ;
* ; ; src o-o send_rtp_sink_%u ; ;
* ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ;
* o----------o sink src o-o sink src o--o sink%u ; '--------------------' ;
* ; '---------------' ,-----------------, '------------' ;
* '-----------------------------------------------------------------------------------------------------------------'
*/
GstPadTemplate *rtp_templ;
GstPad *rtp_sink, *sinkpad, *srcpad;
gchar *pad_name;
WebRTCTransceiver *trans;
GstElement *clocksync;
GstElement *fec_encoder;
g_return_val_if_fail (pad->trans != NULL, NULL);
@ -4398,6 +4476,19 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
srcpad = gst_element_get_static_pad (clocksync, "src");
sinkpad = gst_element_get_static_pad (clocksync, "sink");
if ((fec_encoder = _build_fec_encoder (webrtc, trans))) {
GstPad *fec_sink;
gst_bin_add (GST_BIN (webrtc), fec_encoder);
gst_element_sync_state_with_parent (fec_encoder);
fec_sink = gst_element_get_static_pad (fec_encoder, "sink");
gst_pad_link (srcpad, fec_sink);
gst_object_unref (srcpad);
gst_object_unref (fec_sink);
srcpad = gst_element_get_static_pad (fec_encoder, "src");
}
if (!webrtc->rtpfunnel) {
rtp_templ =
_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
@ -4681,6 +4772,7 @@ _update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
}
item.pt = pt;
item.media_idx = media_idx;
gst_caps_unref (outcaps);
g_array_append_val (stream->ptmap, item);
@ -6359,7 +6451,7 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
stream = _find_transport_for_session (webrtc, session_id);
if (stream)
have_rtx = transport_stream_get_pt (stream, "RTX") != 0;
have_rtx = transport_stream_get_pt (stream, "RTX", -1) != 0;
GST_LOG_OBJECT (webrtc, "requesting aux sender for stream %" GST_PTR_FORMAT
" with pt map %" GST_PTR_FORMAT, stream, pt_map);
@ -6428,20 +6520,43 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
GstElement *prev = NULL;
GstPad *sinkpad = NULL;
TransportStream *stream;
gint red_pt = 0;
gint rtx_pt = 0;
GValue red_pt_array = { 0, };
gboolean have_red_pt = FALSE;
g_value_init (&red_pt_array, GST_TYPE_ARRAY);
stream = _find_transport_for_session (webrtc, session_id);
if (stream) {
red_pt = transport_stream_get_pt (stream, "RED");
rtx_pt = transport_stream_get_pt (stream, "RTX");
guint i = 0;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
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"), "RED")) {
GValue ptval = { 0, };
g_value_init (&ptval, G_TYPE_INT);
g_value_set_int (&ptval, item->pt);
gst_value_array_append_value (&red_pt_array, &ptval);
g_value_unset (&ptval);
have_red_pt = TRUE;
}
}
}
rtx_pt = transport_stream_get_pt (stream, "RTX", -1);
}
GST_LOG_OBJECT (webrtc, "requesting aux receiver for stream %" GST_PTR_FORMAT,
stream);
if (red_pt || rtx_pt)
if (have_red_pt || rtx_pt)
ret = gst_bin_new (NULL);
if (rtx_pt) {
@ -6461,15 +6576,14 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
prev = gst_object_ref (stream->rtxreceive);
}
if (red_pt) {
if (have_red_pt) {
GstElement *rtpreddec = gst_element_factory_make ("rtpreddec", NULL);
GST_DEBUG_OBJECT (webrtc, "Creating RED decoder for pt %d in session %u",
red_pt, session_id);
GST_DEBUG_OBJECT (webrtc, "Creating RED decoder in session %u", session_id);
gst_bin_add (GST_BIN (ret), rtpreddec);
g_object_set (rtpreddec, "pt", red_pt, NULL);
g_object_set_property (G_OBJECT (rtpreddec), "payloads", &red_pt_array);
if (prev)
gst_element_link (prev, rtpreddec);
@ -6497,6 +6611,7 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
}
out:
g_value_unset (&red_pt_array);
return ret;
error:
@ -6506,12 +6621,12 @@ error:
}
static GstElement *
on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
GstWebRTCBin * webrtc)
on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id,
guint ssrc, guint pt, GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstElement *ret = NULL;
gint pt = 0;
gint fec_pt = 0;
GObject *internal_storage;
stream = _find_transport_for_session (webrtc, session_id);
@ -6526,105 +6641,33 @@ on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
* we detect it (by connecting to ptdemux:new-payload-type for
* example)
*/
if (stream)
pt = transport_stream_get_pt (stream, "ULPFEC");
if (stream) {
guint i;
if (pt) {
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (item->pt == pt) {
fec_pt = transport_stream_get_pt (stream, "ULPFEC", item->media_idx);
break;
}
}
}
if (fec_pt) {
GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
pt, session_id);
fec_pt, session_id);
ret = gst_element_factory_make ("rtpulpfecdec", NULL);
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
&internal_storage);
g_object_set (ret, "pt", pt, "storage", internal_storage, NULL);
g_object_set (ret, "pt", fec_pt, "storage", internal_storage, NULL);
g_object_unref (internal_storage);
}
return ret;
}
static GstElement *
on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
GstWebRTCBin * webrtc)
{
GstElement *ret = NULL;
GstElement *prev = NULL;
TransportStream *stream;
guint ulpfec_pt = 0;
guint red_pt = 0;
GstPad *sinkpad = NULL;
GstWebRTCRTPTransceiver *trans;
stream = _find_transport_for_session (webrtc, session_id);
trans = _find_transceiver (webrtc, &session_id,
(FindTransceiverFunc) transceiver_match_for_mline);
if (stream) {
ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC");
red_pt = transport_stream_get_pt (stream, "RED");
}
if (ulpfec_pt || red_pt)
ret = gst_bin_new (NULL);
if (ulpfec_pt) {
GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
GstCaps *caps = transport_stream_get_caps_for_pt (stream, ulpfec_pt);
GST_DEBUG_OBJECT (webrtc,
"Creating ULPFEC encoder for session %d with pt %d", session_id,
ulpfec_pt);
gst_bin_add (GST_BIN (ret), fecenc);
sinkpad = gst_element_get_static_pad (fecenc, "sink");
g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
WEBRTC_TRANSCEIVER (trans)->fec_percentage, NULL);
if (caps && !gst_caps_is_empty (caps)) {
const GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *media = gst_structure_get_string (s, "media");
if (!g_strcmp0 (media, "video"))
g_object_set (fecenc, "multipacket", TRUE, NULL);
}
prev = fecenc;
}
if (red_pt) {
GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for session %d with pt %d",
session_id, red_pt);
gst_bin_add (GST_BIN (ret), redenc);
if (prev)
gst_element_link (prev, redenc);
else
sinkpad = gst_element_get_static_pad (redenc, "sink");
g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
prev = redenc;
}
if (sinkpad) {
GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
gst_object_unref (sinkpad);
gst_element_add_pad (ret, ghost);
}
if (prev) {
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
gst_object_unref (srcpad);
gst_element_add_pad (ret, ghost);
}
return ret;
}
static void
on_rtpbin_bye_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
@ -6788,10 +6831,8 @@ _create_rtpbin (GstWebRTCBin * webrtc)
G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc);
g_signal_connect (rtpbin, "new-storage",
G_CALLBACK (on_rtpbin_new_storage), webrtc);
g_signal_connect (rtpbin, "request-fec-decoder",
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",

7
subprojects/gst-plugins-bad/ext/webrtc/transportstream.c
View file

@ -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"),

4
subprojects/gst-plugins-bad/ext/webrtc/transportstream.h
View file

@ -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);