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c16d4d2c33
Without this it might happen that received data from the DTLS transport is already passed to sctpdec before its state was set to PLAYING. This would cause the data to be dropped, GST_FLOW_FLUSHING to be returned and the whole DTLS transport to shut down. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1172 among other things.
66 lines
2.5 KiB
C
66 lines
2.5 KiB
C
/* GStreamer
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* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WEBRTC_SCTP_TRANSPORT_H__
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#define __GST_WEBRTC_SCTP_TRANSPORT_H__
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#include <gst/gst.h>
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/* libnice */
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#include <agent.h>
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#include <gst/webrtc/webrtc.h>
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#include "gstwebrtcice.h"
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G_BEGIN_DECLS
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GType gst_webrtc_sctp_transport_get_type(void);
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#define GST_TYPE_WEBRTC_SCTP_TRANSPORT (gst_webrtc_sctp_transport_get_type())
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#define GST_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
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#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
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#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
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#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
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#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
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struct _GstWebRTCSCTPTransport
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{
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GstObject parent;
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GstWebRTCDTLSTransport *transport;
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GstWebRTCSCTPTransportState state;
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guint64 max_message_size;
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guint max_channels;
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gboolean association_established;
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gulong sctpdec_block_id;
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GstElement *sctpdec;
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GstElement *sctpenc;
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GstWebRTCBin *webrtcbin;
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};
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struct _GstWebRTCSCTPTransportClass
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{
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GstObjectClass parent_class;
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};
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GstWebRTCSCTPTransport * gst_webrtc_sctp_transport_new (void);
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G_END_DECLS
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#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
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