gstreamer/ext/webrtc/gstwebrtcice.h
Chris Ayoup 9937101e51 webrtc: move filtering properties to webrtcice
We want webrtcbin to only expose properties that are defined in JSEP, so
these additional properties should be moved out.  In order to access
them, the webrtcice instance is exposed from webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00

86 lines
4.1 KiB
C

/* GStreamer
* Copyright (C) 2017 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 __GST_WEBRTC_ICE_H__
#define __GST_WEBRTC_ICE_H__
#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#include <gst/webrtc/webrtc.h>
#include "fwd.h"
G_BEGIN_DECLS
#define GST_WEBRTC_ICE_ERROR gst_webrtc_ice_error_quark ()
GQuark gst_webrtc_ice_error_quark (void);
GType gst_webrtc_ice_get_type(void);
#define GST_TYPE_WEBRTC_ICE (gst_webrtc_ice_get_type())
#define GST_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ICE,GstWebRTCICE))
#define GST_IS_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ICE))
#define GST_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass))
#define GST_IS_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE))
#define GST_WEBRTC_ICE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass))
struct _GstWebRTCICE
{
GstObject parent;
GstWebRTCICEGatheringState ice_gathering_state;
GstWebRTCICEConnectionState ice_connection_state;
GstUri *stun_server;
GstUri *turn_server;
GHashTable *turn_servers;
GstWebRTCICEPrivate *priv;
};
struct _GstWebRTCICEClass
{
GstObjectClass parent_class;
};
GstWebRTCICE * gst_webrtc_ice_new (const gchar * name);
GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
guint session_id);
GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
GstWebRTCICEComponent component);
gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
/* FIXME: GstStructure-ize the candidate */
void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
const gchar * candidate);
gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
const gchar * uri);
G_END_DECLS
#endif /* __GST_WEBRTC_ICE_H__ */