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webrtcbin: Split pad name from mline

Browse source 
The simple case where this breaks is if you add a
datachannel and want to add a new pad (a new media) after). Another
case where this is broken is if the order of the media is forced to
something different by the peer.

It's more simple to just split both things completely. In practice, the
pads will be named in the order in which they are allocated, so it
shouldn't change the current behaviour, just enable new ones.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
This commit is contained in:
Olivier Crête 2021-03-23 17:51:16 -04:00
parent bd9f675318
commit b6114a7fed
2 changed files with 45 additions and 31 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

74
ext/webrtc/gstwebrtcbin.c
View file

@ -636,21 +636,21 @@ _remove_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
typedef struct
{
GstPadDirection direction;
guint mlineindex;
guint mline;
} MLineMatch;
static gboolean
pad_match_for_mline (GstWebRTCBinPad * pad, const MLineMatch * match)
{
return GST_PAD_DIRECTION (pad) == match->direction
&& pad->mlineindex == match->mlineindex;
&& pad->trans->mline == match->mline;
}
static GstWebRTCBinPad *
_find_pad_for_mline (GstWebRTCBin * webrtc, GstPadDirection direction,
guint mlineindex)
guint mline)
{
MLineMatch m = { direction, mlineindex };
MLineMatch m = { direction, mline };
return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_mline);
}
@ -2991,6 +2991,9 @@ _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
}
}
webrtc->priv->max_sink_pad_serial = MAX (webrtc->priv->max_sink_pad_serial,
media_idx);
g_assert (media_idx == gst_sdp_message_medias_len (ret));
if (bundled_mids) {
@ -3722,17 +3725,25 @@ gst_webrtc_bin_create_answer (GstWebRTCBin * webrtc,
static GstWebRTCBinPad *
_create_pad_for_sdp_media (GstWebRTCBin * webrtc, GstPadDirection direction,
guint media_idx)
GstWebRTCRTPTransceiver * trans, guint serial)
{
GstWebRTCBinPad *pad;
gchar *pad_name;
if (direction == GST_PAD_SINK) {
if (serial == G_MAXUINT)
serial = webrtc->priv->max_sink_pad_serial++;
} else {
serial = trans->mline;
}
pad_name =
g_strdup_printf ("%s_%u", direction == GST_PAD_SRC ? "src" : "sink",
media_idx);
serial);
pad = gst_webrtc_bin_pad_new (pad_name, direction);
g_free (pad_name);
pad->mlineindex = media_idx;
pad->trans = gst_object_ref (trans);
return pad;
}
@ -3804,10 +3815,10 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
g_return_val_if_fail (pad->trans != NULL, NULL);
GST_INFO_OBJECT (pad, "linking input stream %u", pad->mlineindex);
trans = WEBRTC_TRANSCEIVER (pad->trans);
GST_INFO_OBJECT (pad, "linking input stream %u", pad->trans->mline);
g_assert (trans->stream);
if (!webrtc->rtpfunnel) {
@ -3816,20 +3827,20 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
"send_rtp_sink_%u");
g_assert (rtp_templ);
pad_name = g_strdup_printf ("send_rtp_sink_%u", pad->mlineindex);
pad_name = g_strdup_printf ("send_rtp_sink_%u", pad->trans->mline);
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_object_unref (rtp_sink);
pad_name = g_strdup_printf ("send_rtp_src_%u", pad->mlineindex);
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"))
g_warn_if_reached ();
g_free (pad_name);
} else {
gchar *pad_name = g_strdup_printf ("sink_%u", pad->mlineindex);
gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline);
GstPad *funnel_sinkpad =
gst_element_get_request_pad (webrtc->rtpfunnel, pad_name);
@ -4249,18 +4260,16 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ||
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *pad =
_find_pad_for_mline (webrtc, GST_PAD_SINK, media_idx);
_find_pad_for_transceiver (webrtc, GST_PAD_SINK, rtp_trans);
if (pad) {
GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT, pad, trans);
g_assert (pad->trans == rtp_trans);
g_assert (pad->mlineindex == media_idx);
gst_object_unref (pad);
} else {
GST_DEBUG_OBJECT (webrtc,
"creating new send pad for transceiver %" GST_PTR_FORMAT, trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, media_idx);
pad->trans = gst_object_ref (rtp_trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, rtp_trans,
G_MAXUINT);
_connect_input_stream (webrtc, pad);
_add_pad (webrtc, pad);
}
@ -4268,18 +4277,16 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY ||
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *pad =
_find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx);
_find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
if (pad) {
GST_DEBUG_OBJECT (webrtc, "found existing receive pad %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT, pad, trans);
g_assert (pad->trans == rtp_trans);
g_assert (pad->mlineindex == media_idx);
gst_object_unref (pad);
} else {
GST_DEBUG_OBJECT (webrtc,
"creating new receive pad for transceiver %" GST_PTR_FORMAT, trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, media_idx);
pad->trans = gst_object_ref (rtp_trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans,
G_MAXUINT);
if (!trans->stream) {
TransportStream *item;
@ -4892,13 +4899,13 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
continue;
}
if (pad->mlineindex >= gst_sdp_message_medias_len (sd->sdp->sdp)) {
if (pad->trans->mline >= gst_sdp_message_medias_len (sd->sdp->sdp)) {
GST_DEBUG_OBJECT (pad, "not mentioned in this description. Skipping");
tmp = tmp->next;
continue;
}
media = gst_sdp_message_get_media (sd->sdp->sdp, pad->mlineindex);
media = gst_sdp_message_get_media (sd->sdp->sdp, pad->trans->mline);
/* skip rejected media */
if (gst_sdp_media_get_port (media) == 0) {
/* FIXME: arrange for an appropriate flow return */
@ -6193,6 +6200,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
if (templ->direction == GST_PAD_SINK ||
g_strcmp0 (templ->name_template, "sink_%u") == 0) {
GstWebRTCRTPTransceiver *trans;
GstWebRTCBinPad *pad2;
GST_OBJECT_LOCK (webrtc);
if (name == NULL || strlen (name) < 6 || !g_str_has_prefix (name, "sink_")) {
@ -6206,19 +6214,27 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
}
GST_OBJECT_UNLOCK (webrtc);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, serial);
trans = _find_transceiver_for_mline (webrtc, serial);
/* Ignore transceivers that already have a pad allocated */
pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, trans);
if (pad2) {
serial = -1;
trans = NULL;
gst_object_unref (pad2);
}
if (!trans) {
trans =
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, serial));
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
" for mline %u", trans, serial);
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1));
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT,
trans);
} else {
GST_LOG_OBJECT (webrtc, "Using existing transceiver %" GST_PTR_FORMAT
" for mline %u", trans, serial);
}
pad->trans = gst_object_ref (trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, trans, serial);
if (caps && name && !_update_transceiver_kind_from_caps (trans, caps))
GST_WARNING_OBJECT (webrtc,

2
ext/webrtc/gstwebrtcbin.h
View file

@ -42,8 +42,6 @@ struct _GstWebRTCBinPad
{
GstGhostPad parent;
guint mlineindex;
GstWebRTCRTPTransceiver *trans;
gulong block_id;