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webrtc: Split sctptransport into lib and implementation parts

Browse source 
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>
This commit is contained in:
Johan Sternerup 2021-05-07 08:12:25 +02:00 committed by GStreamer Marge Bot
parent 7f9bb15055
commit 607ef6db60
12 changed files with 236 additions and 119 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

12
ext/webrtc/gstwebrtcbin.c
View file

@ -29,7 +29,7 @@
#include "webrtcsdp.h"
#include "webrtctransceiver.h"
#include "webrtcdatachannel.h"
#include "sctptransport.h"
#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
@ -1986,7 +1986,7 @@ gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
/* If one stream has a non-default priority, then everyone else does too */
gst_webrtc_bin_attach_tos (webrtc);
gst_webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
sctp_priority);
}
@ -2201,7 +2201,7 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
}
static void
_on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec,
GstWebRTCBin * webrtc)
{
GstWebRTCSCTPTransportState state;
@ -2238,7 +2238,7 @@ _sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused)
TransportStream *stream;
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransportState dtls_state;
GstWebRTCSCTPTransport *sctp_transport;
WebRTCSCTPTransport *sctp_transport;
stream = webrtc->priv->data_channel_transport;
transport = stream->transport;
@ -2326,7 +2326,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
{
if (!webrtc->priv->data_channel_transport) {
TransportStream *stream;
GstWebRTCSCTPTransport *sctp_transport;
WebRTCSCTPTransport *sctp_transport;
stream = _find_transport_for_session (webrtc, session_id);
@ -2336,7 +2336,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
webrtc->priv->data_channel_transport = stream;
if (!(sctp_transport = webrtc->priv->sctp_transport)) {
sctp_transport = gst_webrtc_sctp_transport_new ();
sctp_transport = webrtc_sctp_transport_new ();
sctp_transport->transport =
g_object_ref (webrtc->priv->data_channel_transport->transport);
sctp_transport->webrtcbin = webrtc;

3
ext/webrtc/gstwebrtcbin.h
View file

@ -24,6 +24,7 @@
#include "fwd.h"
#include "gstwebrtcice.h"
#include "transportstream.h"
#include "webrtcsctptransport.h"
G_BEGIN_DECLS
@ -106,7 +107,7 @@ struct _GstWebRTCBinPrivate
guint jb_latency;
GstWebRTCSCTPTransport *sctp_transport;
WebRTCSCTPTransport *sctp_transport;
TransportStream *data_channel_transport;
GstWebRTCICE *ice;

2
ext/webrtc/meson.build
View file

@ -4,7 +4,7 @@ webrtc_sources = [
'gstwebrtcstats.c',
'icestream.c',
'nicetransport.c',
'sctptransport.c',
'webrtcsctptransport.c',
'gstwebrtcbin.c',
'transportreceivebin.c',
'transportsendbin.c',

6
ext/webrtc/webrtcdatachannel.c
View file

@ -356,7 +356,7 @@ _close_procedure (WebRTCDataChannel * channel, gpointer user_data)
}
static void
_on_sctp_stream_reset (GstWebRTCSCTPTransport * sctp, guint stream_id,
_on_sctp_stream_reset (WebRTCSCTPTransport * sctp, guint stream_id,
WebRTCDataChannel * channel)
{
if (channel->parent.id == stream_id) {
@ -1003,7 +1003,7 @@ webrtc_data_channel_init (WebRTCDataChannel * channel)
static void
_data_channel_set_sctp_transport (WebRTCDataChannel * channel,
GstWebRTCSCTPTransport * sctp)
WebRTCSCTPTransport * sctp)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
g_return_if_fail (GST_IS_WEBRTC_SCTP_TRANSPORT (sctp));
@ -1026,7 +1026,7 @@ _data_channel_set_sctp_transport (WebRTCDataChannel * channel,
void
webrtc_data_channel_link_to_sctp (WebRTCDataChannel * channel,
GstWebRTCSCTPTransport * sctp_transport)
WebRTCSCTPTransport * sctp_transport)
{
if (sctp_transport && !channel->sctp_transport) {
gint id;

6
ext/webrtc/webrtcdatachannel.h
View file

@ -24,7 +24,7 @@
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/dtlstransport.h>
#include <gst/webrtc/datachannel.h>
#include "sctptransport.h"
#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
@ -45,7 +45,7 @@ struct _WebRTCDataChannel
{
GstWebRTCDataChannel parent;
GstWebRTCSCTPTransport *sctp_transport;
WebRTCSCTPTransport *sctp_transport;
GstElement *appsrc;
GstElement *appsink;
@ -68,7 +68,7 @@ struct _WebRTCDataChannelClass
void webrtc_data_channel_start_negotiation (WebRTCDataChannel *channel);
G_GNUC_INTERNAL
void webrtc_data_channel_link_to_sctp (WebRTCDataChannel *channel,
GstWebRTCSCTPTransport *sctp_transport);
WebRTCSCTPTransport *sctp_transport);
G_END_DECLS

111
ext/webrtc/sctptransport.c → ext/webrtc/webrtcsctptransport.c
View file

@ -23,10 +23,10 @@
#include <stdio.h>
#include "sctptransport.h"
#include "webrtcsctptransport.h"
#include "gstwebrtcbin.h"
#define GST_CAT_DEFAULT gst_webrtc_sctp_transport_debug
#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
@ -45,18 +45,19 @@ enum
PROP_MAX_CHANNELS,
};
static guint gst_webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
static guint webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
#define gst_webrtc_sctp_transport_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_sctp_transport_debug,
#define webrtc_sctp_transport_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (WebRTCSCTPTransport, webrtc_sctp_transport,
GST_TYPE_WEBRTC_SCTP_TRANSPORT,
GST_DEBUG_CATEGORY_INIT (webrtc_sctp_transport_debug,
"webrtcsctptransport", 0, "webrtcsctptransport"););
typedef void (*SCTPTask) (GstWebRTCSCTPTransport * sctp, gpointer user_data);
typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
struct task
{
GstWebRTCSCTPTransport *sctp;
WebRTCSCTPTransport *sctp;
SCTPTask func;
gpointer user_data;
GDestroyNotify notify;
@ -81,7 +82,7 @@ _free_task (struct task *task)
}
static void
_sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func,
_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
gpointer user_data, GDestroyNotify notify)
{
struct task *task = g_new0 (struct task, 1);
@ -97,17 +98,17 @@ _sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func,
}
static void
_emit_stream_reset (GstWebRTCSCTPTransport * sctp, gpointer user_data)
_emit_stream_reset (WebRTCSCTPTransport * sctp, gpointer user_data)
{
guint stream_id = GPOINTER_TO_UINT (user_data);
g_signal_emit (sctp,
gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
}
static void
_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
GstWebRTCSCTPTransport * sctp)
WebRTCSCTPTransport * sctp)
{
guint stream_id;
@ -120,7 +121,7 @@ _on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
static void
_on_sctp_association_established (GstElement * sctpenc, gboolean established,
GstWebRTCSCTPTransport * sctp)
WebRTCSCTPTransport * sctp)
{
GST_OBJECT_LOCK (sctp);
if (established)
@ -133,21 +134,8 @@ _on_sctp_association_established (GstElement * sctpenc, gboolean established,
g_object_notify (G_OBJECT (sctp), "state");
}
static void
gst_webrtc_sctp_transport_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
// GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
void
gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp,
webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
GstWebRTCPriorityType priority)
{
GstPad *pad;
@ -161,10 +149,10 @@ gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp,
}
static void
gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
switch (prop_id) {
case PROP_TRANSPORT:
@ -186,9 +174,9 @@ gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
}
static void
gst_webrtc_sctp_transport_finalize (GObject * object)
webrtc_sctp_transport_finalize (GObject * object)
{
GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
g_signal_handlers_disconnect_by_data (sctp->sctpdec, sctp);
g_signal_handlers_disconnect_by_data (sctp->sctpenc, sctp);
@ -202,9 +190,9 @@ gst_webrtc_sctp_transport_finalize (GObject * object)
}
static void
gst_webrtc_sctp_transport_constructed (GObject * object)
webrtc_sctp_transport_constructed (GObject * object)
{
GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
guint association_id;
association_id = g_random_int_range (0, G_MAXUINT16);
@ -226,61 +214,38 @@ gst_webrtc_sctp_transport_constructed (GObject * object)
}
static void
gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = gst_webrtc_sctp_transport_constructed;
gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
gobject_class->set_property = gst_webrtc_sctp_transport_set_property;
gobject_class->finalize = gst_webrtc_sctp_transport_finalize;
gobject_class->constructed = webrtc_sctp_transport_constructed;
gobject_class->get_property = webrtc_sctp_transport_get_property;
gobject_class->finalize = webrtc_sctp_transport_finalize;
g_object_class_install_property (gobject_class,
PROP_TRANSPORT,
g_param_spec_object ("transport",
"WebRTC DTLS Transport",
"DTLS transport used for this SCTP transport",
GST_TYPE_WEBRTC_DTLS_TRANSPORT,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_STATE,
g_param_spec_enum ("state",
"WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_MESSAGE_SIZE,
g_param_spec_uint64 ("max-message-size",
"Maximum message size",
"Maximum message size as reported by the transport", 0, G_MAXUINT64,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_CHANNELS,
g_param_spec_uint ("max-channels",
"Maximum number of channels", "Maximum number of channels",
0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_override_property (gobject_class, PROP_TRANSPORT, "transport");
g_object_class_override_property (gobject_class, PROP_STATE, "state");
g_object_class_override_property (gobject_class,
PROP_MAX_MESSAGE_SIZE, "max-message-size");
g_object_class_override_property (gobject_class,
PROP_MAX_CHANNELS, "max-channels");
/**
* GstWebRTCSCTPTransport::stream-reset:
* @object: the #GstWebRTCSCTPTransport
* WebRTCSCTPTransport::stream-reset:
* @object: the #WebRTCSCTPTransport
* @stream_id: the SCTP stream that was reset
*/
gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
g_signal_new ("stream-reset", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
}
static void
gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
{
}
GstWebRTCSCTPTransport *
gst_webrtc_sctp_transport_new (void)
WebRTCSCTPTransport *
webrtc_sctp_transport_new (void)
{
return g_object_new (GST_TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
return g_object_new (TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
}

74
ext/webrtc/webrtcsctptransport.h Normal file
View file

@ -0,0 +1,74 @@
/* GStreamer
* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __WEBRTC_SCTP_TRANSPORT_H__
#define __WEBRTC_SCTP_TRANSPORT_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc.h>
#include <gst/webrtc/sctptransport.h>
#include "gstwebrtcice.h"
#include "gst/webrtc/webrtc-priv.h"
G_BEGIN_DECLS
GType webrtc_sctp_transport_get_type(void);
#define TYPE_WEBRTC_SCTP_TRANSPORT (webrtc_sctp_transport_get_type())
#define WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransport))
#define WEBRTC_IS_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),TYPE_WEBRTC_SCTP_TRANSPORT))
#define WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
#define WEBRTC_SCTP_IS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT))
#define WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
typedef struct _WebRTCSCTPTransport WebRTCSCTPTransport;
typedef struct _WebRTCSCTPTransportClass WebRTCSCTPTransportClass;
struct _WebRTCSCTPTransport
{
GstWebRTCSCTPTransport parent;
GstWebRTCDTLSTransport *transport;
GstWebRTCSCTPTransportState state;
guint64 max_message_size;
guint max_channels;
gboolean association_established;
gulong sctpdec_block_id;
GstElement *sctpdec;
GstElement *sctpenc;
GstWebRTCBin *webrtcbin;
};
struct _WebRTCSCTPTransportClass
{
GstWebRTCSCTPTransportClass parent_class;
};
WebRTCSCTPTransport * webrtc_sctp_transport_new (void);
void
webrtc_sctp_transport_set_priority (WebRTCSCTPTransport *sctp,
GstWebRTCPriorityType priority);
G_END_DECLS
#endif /* __WEBRTC_SCTP_TRANSPORT_H__ */

2
gst-libs/gst/webrtc/meson.build
View file

@ -6,6 +6,7 @@ webrtc_sources = [
'rtpsender.c',
'rtptransceiver.c',
'datachannel.c',
'sctptransport.c',
]
webrtc_headers = [
@ -18,6 +19,7 @@ webrtc_headers = [
'datachannel.h',
'webrtc_fwd.h',
'webrtc.h',
'sctptransport.h',
]
webrtc_enumtypes_headers = [

79
gst-libs/gst/webrtc/sctptransport.c Normal file
View file

@ -0,0 +1,79 @@
/* GStreamer
* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "sctptransport.h"
#include "webrtc-priv.h"
G_DEFINE_ABSTRACT_TYPE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
GST_TYPE_OBJECT);
static void
gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
/* all properties should by handled by the plugin class */
g_assert_not_reached ();
}
static void
gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
guint property_id_dummy = 0;
gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
g_object_class_install_property (gobject_class,
++property_id_dummy,
g_param_spec_object ("transport",
"WebRTC DTLS Transport",
"DTLS transport used for this SCTP transport",
GST_TYPE_WEBRTC_DTLS_TRANSPORT,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
++property_id_dummy,
g_param_spec_enum ("state",
"WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
++property_id_dummy,
g_param_spec_uint64 ("max-message-size",
"Maximum message size",
"Maximum message size as reported by the transport", 0, G_MAXUINT64,
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
++property_id_dummy,
g_param_spec_uint ("max-channels",
"Maximum number of channels", "Maximum number of channels",
0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
{
}

36
ext/webrtc/sctptransport.h → gst-libs/gst/webrtc/sctptransport.h
View file

@ -21,14 +21,13 @@
#define __GST_WEBRTC_SCTP_TRANSPORT_H__
#include <gst/gst.h>
/* libnice */
#include <agent.h>
#include <gst/webrtc/webrtc.h>
#include "gstwebrtcice.h"
#include <gst/webrtc/webrtc_fwd.h>
G_BEGIN_DECLS
GST_WEBRTC_API
GType gst_webrtc_sctp_transport_get_type(void);
#define GST_TYPE_WEBRTC_SCTP_TRANSPORT (gst_webrtc_sctp_transport_get_type())
#define GST_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
@ -36,34 +35,7 @@ GType gst_webrtc_sctp_transport_get_type(void);
#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
struct _GstWebRTCSCTPTransport
{
GstObject parent;
GstWebRTCDTLSTransport *transport;
GstWebRTCSCTPTransportState state;
guint64 max_message_size;
guint max_channels;
gboolean association_established;
gulong sctpdec_block_id;
GstElement *sctpdec;
GstElement *sctpenc;
GstWebRTCBin *webrtcbin;
};
struct _GstWebRTCSCTPTransportClass
{
GstObjectClass parent_class;
};
GstWebRTCSCTPTransport * gst_webrtc_sctp_transport_new (void);
void
gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
GstWebRTCPriorityType priority);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSCTPTransport, gst_object_unref)
G_END_DECLS

21
gst-libs/gst/webrtc/webrtc-priv.h
View file

@ -289,6 +289,27 @@ GST_WEBRTC_API
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
/**
* GstWebRTCSCTPTransport:
*
* Since: 1.20
*/
struct _GstWebRTCSCTPTransport
{
GstObject parent;
};
/**
* GstWebRTCSCTPTransportClass:
*
* Since: 1.20
*/
struct _GstWebRTCSCTPTransportClass
{
GstObjectClass parent_class;
};
G_END_DECLS
#endif /* __GST_WEBRTC_PRIV_H__ */

3
gst-libs/gst/webrtc/webrtc_fwd.h
View file

@ -92,6 +92,9 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
/**
* GstWebRTCDTLSTransportState:
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new