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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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607ef6db60
GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
79 lines
2.7 KiB
C
79 lines
2.7 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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "sctptransport.h"
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#include "webrtc-priv.h"
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G_DEFINE_ABSTRACT_TYPE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
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GST_TYPE_OBJECT);
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static void
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gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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/* all properties should by handled by the plugin class */
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g_assert_not_reached ();
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}
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static void
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gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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guint property_id_dummy = 0;
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gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
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g_object_class_install_property (gobject_class,
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++property_id_dummy,
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g_param_spec_object ("transport",
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"WebRTC DTLS Transport",
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"DTLS transport used for this SCTP transport",
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GST_TYPE_WEBRTC_DTLS_TRANSPORT,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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++property_id_dummy,
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g_param_spec_enum ("state",
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"WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
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GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
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GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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++property_id_dummy,
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g_param_spec_uint64 ("max-message-size",
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"Maximum message size",
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"Maximum message size as reported by the transport", 0, G_MAXUINT64,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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++property_id_dummy,
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g_param_spec_uint ("max-channels",
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"Maximum number of channels", "Maximum number of channels",
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0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
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{
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}
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