72433cb942
We now fill the foundation, related-address, related-port, username-fragment and tcp-type fields. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8792>
435 lines
17 KiB
C
435 lines
17 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/webrtc/webrtc_fwd.h>
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G_BEGIN_DECLS
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GST_WEBRTC_API
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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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/*< protected >*/
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guint min_rtp_port;
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guint max_rtp_port;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstWebRTCICECandidateStats:
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* @ipaddr: A string containing the address of the candidate. This value may be
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* an IPv4 address, an IPv6 address, or a fully-qualified domain name
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* @port: The network port number used by the candidate
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* @stream_id: A string that uniquely identifies the object that is being
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* monitored to produce this set of statistics
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* @type: The candidate type
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* @proto: A string specifying the protocol (tcp or udp) used to transmit data
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* on the @port
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* @replay_proto: A string identifying the protocol used by the endpoint for
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* communicating with the TURN server; valid values are tcp, udp, and tls
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* @prio: The candidate's priority, corresponding to RTCIceCandidate.priority
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* @url: For local candidates, the url property is the URL of the ICE server
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* from which the candidate was received
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* @foundation: The ICE foundation as defined in RFC5245 section 15.1 (Since: 1.28)
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* @related_address: The ICE rel-addr defined in RFC5245 section 15.1 Only
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* set for serverreflexive, peerreflexive and relay candidates. (Since: 1.28)
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* @related_port: The ICE rel-addr defined in RFC5245 section 15.1. Only set
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* for serverreflexive, peerreflexive and relay candidates. (Since: 1.28)
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* @username_fragment: The ICE username fragment as defined in RFC5245 section 7.1.2.3 (Since: 1.28)
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* @tcp_type: The ICE candidate TCP type, (Since: 1.28)
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*
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* Since: 1.22
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*/
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struct _GstWebRTCICECandidateStats
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{
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gchar *ipaddr;
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guint port;
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guint stream_id;
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const gchar *type;
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const gchar *proto;
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const gchar *relay_proto;
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guint prio;
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gchar *url;
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/**
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* GstWebRTCICECandidateStats.ABI: (attributes doc.skip=true)
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*
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* ABI compatibility union
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*
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* Since: 1.28
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*/
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union {
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/**
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* GstWebRTCICECandidateStats.ABI.abi: (attributes doc.skip=true)
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*
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* ABI compatibility struct
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*
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* Since: 1.28
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*/
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struct {
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/**
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* GstWebRTCICECandidateStats.ABI.abi.foundation:
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*
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* The foundation of the ICE candidate.
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*
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* Since: 1.28
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*/
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gchar *foundation;
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/**
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* GstWebRTCICECandidateStats.ABI.abi.related_address:
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*
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* The related address (STUN or TURN server) of the candidate. Will be
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* NULL for host candidates.
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*
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* Since: 1.28
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*/
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gchar *related_address;
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/**
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* GstWebRTCICECandidateStats.ABI.abi.related_port:
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*
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* The related port (STUN or TURN server) of the candidate. Will be
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* 0 for host candidates.
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*
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* Since: 1.28
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*/
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guint related_port;
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/**
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* GstWebRTCICECandidateStats.ABI.abi.username_fragment:
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*
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* The ICE username for this candidate.
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*
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* Since: 1.28
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*/
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gchar *username_fragment;
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/**
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* GstWebRTCICECandidateStats.ABI.abi.tcp_type:
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*
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* The type of TCP candidate. Will be NULL if the candidate is not a TCP
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* candidate.
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*
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* Since: 1.28
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*/
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GstWebRTCICETcpCandidateType tcp_type;
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} abi;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE];
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} ABI;
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};
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_ADDRESS:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_ADDRESS(c) ((c)->ipaddr)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_PORT:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_PORT(c) ((c)->port)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_STREAM_ID:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_STREAM_ID(c) ((c)->stream_id)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_TYPE:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_TYPE(c) ((c)->type)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_PROTOCOL:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_PROTOCOL(c) ((c)->proto)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_RELAY_PROTOCOL:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELAY_PROTOCOL(c) ((c)->relay_proto)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_PRIORITY:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_PRIORITY(c) ((c)->prio)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_URL:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_URL(c) ((c)->url)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_FOUNDATION:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_FOUNDATION(c) ((c)->ABI.abi.foundation)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_ADDRESS:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_ADDRESS(c) ((c)->ABI.abi.related_address)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_PORT:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_RELATED_PORT(c) ((c)->ABI.abi.related_port)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_USERNAME_FRAGMENT:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_USERNAME_FRAGMENT(c) ((c)->ABI.abi.username_fragment)
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/**
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* GST_WEBRTC_ICE_CANDIDATE_STATS_TCP_TYPE:
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*
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* Since: 1.28
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*/
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#define GST_WEBRTC_ICE_CANDIDATE_STATS_TCP_TYPE(c) ((c)->ABI.abi.tcp_type)
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/**
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* GstWebRTCICEOnCandidateFunc:
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* @ice: The #GstWebRTCICE
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* @stream_id: The stream id
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* @candidate: The discovered candidate
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* @user_data: User data that was set by #gst_webrtc_ice_set_on_ice_candidate
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*
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* Callback function to be triggered on discovery of a new candidate
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* Since: 1.22
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*/
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typedef void (*GstWebRTCICEOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, const gchar * candidate, gpointer user_data);
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struct _GstWebRTCICEClass {
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GstObjectClass parent_class;
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GstWebRTCICEStream * (*add_stream) (GstWebRTCICE * ice,
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guint session_id);
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GstWebRTCICETransport * (*find_transport) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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GstWebRTCICEComponent component);
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gboolean (*gather_candidates) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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void (*add_candidate) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * candidate,
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GstPromise * promise);
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gboolean (*set_local_credentials) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * ufrag,
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const gchar * pwd);
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gboolean (*set_remote_credentials) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * ufrag,
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const gchar * pwd);
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gboolean (*add_turn_server) (GstWebRTCICE * ice,
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const gchar * uri);
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void (*set_is_controller) (GstWebRTCICE * ice,
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gboolean controller);
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gboolean (*get_is_controller) (GstWebRTCICE * ice);
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void (*set_force_relay) (GstWebRTCICE * ice,
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gboolean force_relay);
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void (*set_stun_server) (GstWebRTCICE * ice,
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const gchar * uri);
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gchar * (*get_stun_server) (GstWebRTCICE * ice);
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void (*set_turn_server) (GstWebRTCICE * ice,
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const gchar * uri);
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gchar * (*get_turn_server) (GstWebRTCICE * ice);
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/**
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* GstWebRTCICEClass::set_http_proxy:
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* @ice: a #GstWebRTCICE
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* @uri: (transfer none): URI of the HTTP proxy of the form
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* http://[username:password@]hostname[:port]
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*
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* Set HTTP Proxy to be used when connecting to TURN server.
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*
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* Since: 1.22
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*/
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void (*set_http_proxy) (GstWebRTCICE * ice,
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const gchar * uri);
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/**
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* GstWebRTCICEClass::get_http_proxy:
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* @ice: a #GstWebRTCICE
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*
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* Get HTTP Proxy to be used when connecting to TURN server.
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*
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* Returns: (transfer full): URI of the HTTP proxy of the form
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* http://[username:password@]hostname[:port]
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*
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* Since: 1.22
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*/
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gchar * (*get_http_proxy) (GstWebRTCICE * ice);
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void (*set_tos) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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guint tos);
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void (*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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GstWebRTCICECandidateStats** (*get_local_candidates)(GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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GstWebRTCICECandidateStats**(*get_remote_candidates)(GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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gboolean (*get_selected_pair) (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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GstWebRTCICECandidateStats ** local_stats,
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GstWebRTCICECandidateStats ** remote_stats);
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_WEBRTC_API
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GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
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guint session_id);
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GST_WEBRTC_API
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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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GST_WEBRTC_API
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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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GST_WEBRTC_API
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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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GstPromise * promise);
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GST_WEBRTC_API
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gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * ufrag,
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const gchar * pwd);
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GST_WEBRTC_API
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gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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const gchar * ufrag,
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const gchar * pwd);
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GST_WEBRTC_API
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gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
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const gchar * uri);
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GST_WEBRTC_API
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void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice,
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gboolean controller);
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GST_WEBRTC_API
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gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice);
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GST_WEBRTC_API
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void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice,
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gboolean force_relay);
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GST_WEBRTC_API
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void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice,
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const gchar * uri);
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GST_WEBRTC_API
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gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice);
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GST_WEBRTC_API
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void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice,
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const gchar * uri);
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GST_WEBRTC_API
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gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice);
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GST_WEBRTC_API
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void gst_webrtc_ice_set_http_proxy (GstWebRTCICE * ice,
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const gchar * uri);
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GST_WEBRTC_API
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gchar * gst_webrtc_ice_get_http_proxy (GstWebRTCICE * ice);
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GST_WEBRTC_API
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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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GST_WEBRTC_API
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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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GST_WEBRTC_API
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GstWebRTCICECandidateStats** gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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GST_WEBRTC_API
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GstWebRTCICECandidateStats** gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream);
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GST_WEBRTC_API
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gboolean gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
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GstWebRTCICEStream * stream,
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GstWebRTCICECandidateStats ** local_stats,
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GstWebRTCICECandidateStats ** remote_stats);
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GST_WEBRTC_API
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void gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats);
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GST_WEBRTC_API
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GType gst_webrtc_ice_candidate_stats_get_type (void);
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICE, gst_object_unref)
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GST_WEBRTC_API
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GstWebRTCICECandidateStats * gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats *stats);
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G_END_DECLS
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#endif /* __GST_WEBRTC_ICE_H__ */
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