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gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h

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webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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/* 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__
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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#include <gst/webrtc/webrtc_fwd.h>
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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G_BEGIN_DECLS
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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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
{
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstObject parent;
webrtcbin: New add-turn-server API It is possible and often desirable to pass multiple ICE relays to libnice agents, the "turn-server" property, while convenient to use from the command line, does not allow that. This adds a new action signal, "add-turn-server" to address that. https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-08-22 17:05:02 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstWebRTCICEGatheringState ice_gathering_state;
GstWebRTCICEConnectionState ice_connection_state;
guint min_rtp_port;
guint max_rtp_port;
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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};
webrtcstats: Improve selected candidate pair stats by adding ICE candidate info The implementation follows w3.org specs: * https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* * https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* Corresponding unit tests are also added. Rebased and updated from https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462 Fixes #1207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
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struct _GstWebRTCICECandidateStats
{
gchar *ipaddr;
guint port;
guint stream_id;
const gchar *type;
const gchar *proto;
const gchar *relay_proto;
guint prio;
gchar *url;
};
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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/**
* GstWebRTCICEOnCandidateFunc:
* @ice: The #GstWebRTCICE
* @stream_id: The stream id
* @candidate: The discovered candidate
* @user_data: User data that was set by #gst_webrtc_ice_set_on_ice_candidate
*
* Callback function to be triggered on discovery of a new candidate
* Since: 1.22
*/
typedef void (*GstWebRTCICEOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, gchar * candidate, gpointer user_data);
struct _GstWebRTCICEClass {
GObjectClass parent_class;
GstWebRTCICEStream * (*add_stream) (GstWebRTCICE * ice,
guint session_id);
GstWebRTCICETransport * (*find_transport) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
GstWebRTCICEComponent component);
gboolean (*gather_candidates) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
void (*add_candidate) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
const gchar * candidate);
gboolean (*set_local_credentials) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean (*set_remote_credentials) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
gboolean (*add_turn_server) (GstWebRTCICE * ice,
const gchar * uri);
void (*set_is_controller) (GstWebRTCICE * ice,
gboolean controller);
gboolean (*get_is_controller) (GstWebRTCICE * ice);
void (*set_force_relay) (GstWebRTCICE * ice,
gboolean force_relay);
void (*set_stun_server) (GstWebRTCICE * ice,
const gchar * uri);
gchar * (*get_stun_server) (GstWebRTCICE * ice);
void (*set_turn_server) (GstWebRTCICE * ice,
const gchar * uri);
gchar * (*get_turn_server) (GstWebRTCICE * ice);
void (*set_tos) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
guint tos);
void (*set_on_ice_candidate) (GstWebRTCICE * ice,
GstWebRTCICEOnCandidateFunc func,
gpointer user_data,
GDestroyNotify notify);
GArray * (*get_local_candidates) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
GArray * (*get_remote_candidates) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
gboolean (*get_selected_pair) (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
GstWebRTCICECandidateStats ** local_stats,
GstWebRTCICECandidateStats ** remote_stats);
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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};
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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GstWebRTCICEStream * gst_webrtc_ice_add_stream (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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guint session_id);
GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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GstWebRTCICETransport * gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstWebRTCICEStream * stream,
GstWebRTCICEComponent component);
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
gboolean gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstWebRTCICEStream * stream);
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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/* FIXME: GstStructure-ize the candidate */
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
void gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstWebRTCICEStream * stream,
const gchar * candidate);
GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
gboolean gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
GST_WEBRTC_API
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
gboolean gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GstWebRTCICEStream * stream,
gchar * ufrag,
gchar * pwd);
GST_WEBRTC_API
webrtcbin: New add-turn-server API It is possible and often desirable to pass multiple ICE relays to libnice agents, the "turn-server" property, while convenient to use from the command line, does not allow that. This adds a new action signal, "add-turn-server" to address that. https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-08-22 17:05:02 +00:00
gboolean gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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const gchar * uri);
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
void gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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gboolean controller);
GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
gboolean gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice);
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
void gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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gboolean force_relay);
GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
void gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
const gchar * uri);
GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
gchar * gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice);
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
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GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
void gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
const gchar * uri);
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
gchar * gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice);
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
void gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GstWebRTCICEOnCandidateFunc func,
gpointer user_data,
GDestroyNotify notify);
webrtc: remove private properties/signals from the now public ice object We don't want to expose all of the webrtcbin internals to the world. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-17 05:14:48 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtc: Set the DSCP markings based on the priority This matches how the WebRTC javascript API works and the Chrome implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-07-08 21:24:36 +00:00
void gst_webrtc_ice_set_tos (GstWebRTCICE * ice,
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GstWebRTCICEStream * stream,
guint tos);
webrtcstats: Improve selected candidate pair stats by adding ICE candidate info The implementation follows w3.org specs: * https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* * https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* Corresponding unit tests are also added. Rebased and updated from https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462 Fixes #1207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2021-06-30 20:01:10 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtcstats: Improve selected candidate pair stats by adding ICE candidate info The implementation follows w3.org specs: * https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* * https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* Corresponding unit tests are also added. Rebased and updated from https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462 Fixes #1207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2021-06-30 20:01:10 +00:00
GArray * gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtcstats: Improve selected candidate pair stats by adding ICE candidate info The implementation follows w3.org specs: * https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* * https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* Corresponding unit tests are also added. Rebased and updated from https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462 Fixes #1207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2021-06-30 20:01:10 +00:00
GArray * gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream);
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtcstats: Improve selected candidate pair stats by adding ICE candidate info The implementation follows w3.org specs: * https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* * https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* Corresponding unit tests are also added. Rebased and updated from https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1462 Fixes #1207 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2021-06-30 20:01:10 +00:00
gboolean gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
GstWebRTCICEStream * stream,
GstWebRTCICECandidateStats ** local_stats,
GstWebRTCICECandidateStats ** remote_stats);
webrtc: Refactor ICECandidateStats freeing logic to a dedicated function Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-03-26 12:36:46 +00:00
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
GST_WEBRTC_API
webrtc: Refactor ICECandidateStats freeing logic to a dedicated function Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1998>
2022-03-26 12:36:46 +00:00
void gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats);
webrtc: Split WebRTCICE into base classes and implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>
2022-04-18 17:55:13 +00:00
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICE, gst_object_unref)
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
G_END_DECLS
#endif /* __GST_WEBRTC_ICE_H__ */
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