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webrtcbin: connect input stream when receiving caps

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.. if a current direction has already been set

When `webrtcbin` has created an offer based on codec_preferences,
it might not have received caps on its sinkpads by the time a
remote description is set, in which case we want to connect the
input stream upon actual reception of the caps instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1233>
This commit is contained in:
Mathieu Duponchelle 2021-10-23 01:54:05 +02:00 committed by GStreamer Marge Bot
parent a9506f20d3
commit be0b5c54fd
1 changed files with 49 additions and 6 deletions
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55
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -112,6 +112,9 @@
*/
static void _update_need_negotiation (GstWebRTCBin * webrtc);
static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
GstWebRTCBinPad * pad);
#define GST_CAT_DEFAULT gst_webrtc_bin_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
@ -274,6 +277,19 @@ gst_webrtc_bin_pad_update_ssrc_event (GstWebRTCBinPad * wpad)
}
}
static GList *
_get_pending_sink_transceiver (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
GList *ret;
for (ret = webrtc->priv->pending_sink_transceivers; ret; ret = ret->next) {
if (ret->data == pad)
break;
}
return ret;
}
static gboolean
gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
@ -301,6 +317,35 @@ gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
gst_structure_get_uint (s, "ssrc", &trans->current_ssrc);
gst_webrtc_bin_pad_update_ssrc_event (wpad);
}
/* A remote description might have been set while the pad hadn't
* yet received caps, delaying the connection of the input stream
*/
PC_LOCK (webrtc);
if (wpad->trans) {
GST_OBJECT_LOCK (wpad->trans);
if (wpad->trans->current_direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY
|| wpad->trans->current_direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GList *pending = _get_pending_sink_transceiver (webrtc, wpad);
if (pending) {
GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with "
"transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT,
wpad->trans, wpad->received_caps);
_connect_input_stream (webrtc, wpad);
gst_pad_remove_probe (GST_PAD (pad), wpad->block_id);
wpad->block_id = 0;
gst_object_unref (pending->data);
webrtc->priv->pending_sink_transceivers =
g_list_delete_link (webrtc->priv->pending_sink_transceivers,
pending);
}
}
GST_OBJECT_UNLOCK (wpad->trans);
}
PC_UNLOCK (webrtc);
} else if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
check_negotiation = TRUE;
}
@ -423,10 +468,8 @@ gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
G_ADD_PRIVATE (GstWebRTCBin)
GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
"webrtcbin element"););
static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
GstWebRTCBinPad * pad);
"webrtcbin element");
);
enum
{
@ -1446,8 +1489,8 @@ _check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
/* If connection has created any RTCDataChannel's, and no m= section has
* been negotiated yet for data, return "true". */
if (webrtc->priv->data_channels->len > 0) {
if (_message_get_datachannel_index (webrtc->current_local_description->
sdp) >= G_MAXUINT) {
if (_message_get_datachannel_index (webrtc->
current_local_description->sdp) >= G_MAXUINT) {
GST_LOG_OBJECT (webrtc,
"no data channel media section and have %u " "transports",
webrtc->priv->data_channels->len);