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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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92626535c7
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored) add 'gathering_started' flag to avoid changing ports after gathering has started validity checks: min port <= max port enforced, error thrown otherwise include tests to ensure port range is being utilized (by @hhardy) Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
112 lines
5.9 KiB
C
112 lines
5.9 KiB
C
/* GStreamer
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* Copyright (C) 2017 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_ICE_H__
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#define __GST_WEBRTC_ICE_H__
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#include <gst/gst.h>
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#include <gst/sdp/sdp.h>
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#include <gst/webrtc/webrtc.h>
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#include "fwd.h"
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G_BEGIN_DECLS
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#define GST_WEBRTC_ICE_ERROR gst_webrtc_ice_error_quark ()
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GQuark gst_webrtc_ice_error_quark (void);
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GType gst_webrtc_ice_get_type(void);
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#define GST_TYPE_WEBRTC_ICE (gst_webrtc_ice_get_type())
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#define GST_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ICE,GstWebRTCICE))
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#define GST_IS_WEBRTC_ICE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_ICE))
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#define GST_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass))
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#define GST_IS_WEBRTC_ICE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE))
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#define GST_WEBRTC_ICE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE,GstWebRTCICEClass))
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struct _GstWebRTCICE
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{
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GstObject parent;
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GstWebRTCICEGatheringState ice_gathering_state;
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GstWebRTCICEConnectionState ice_connection_state;
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GstUri *stun_server;
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GstUri *turn_server;
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GHashTable *turn_servers;
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GstWebRTCICEPrivate *priv;
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guint min_rtp_port;
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guint max_rtp_port;
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};
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struct _GstWebRTCICEClass
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{
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GstObjectClass parent_class;
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};
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GstWebRTCICE * gst_webrtc_ice_new (const gchar * name);
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GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
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guint session_id);
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GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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GstWebRTCICEComponent component);
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gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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/* FIXME: GstStructure-ize the candidate */
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void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * candidate);
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gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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gchar * ufrag,
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gchar * pwd);
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gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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gchar * ufrag,
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gchar * pwd);
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gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
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const gchar * uri);
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void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice,
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gboolean controller);
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gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice);
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void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice,
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gboolean force_relay);
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void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice,
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const gchar * uri);
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gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice);
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void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice,
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const gchar * uri);
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gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice);
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typedef void (*GstWebRTCIceOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, gchar * candidate, gpointer user_data);
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void gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
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GstWebRTCIceOnCandidateFunc func,
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gpointer user_data,
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GDestroyNotify notify);
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void gst_webrtc_ice_set_tos (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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guint tos);
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G_END_DECLS
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#endif /* __GST_WEBRTC_ICE_H__ */
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