gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.c
Thibault Saunier 24b171735d webrtc: Fix documentaton moving symbols in the right pages
As those symbols are documented in a 'fwd' header smart indexing in
hotdoc wasn't working.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2891>
2022-08-18 08:34:36 +00:00

555 lines
15 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.
*/
/**
* SECTION:gstwebrtcice
* @title: GstWebRTCICE
* @short_description: Base class WebRTC ICE handling
* @symbols:
* - GstWebRTCICE
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "ice.h"
#include "icestream.h"
#include "webrtc-priv.h"
#define GST_CAT_DEFAULT gst_webrtc_ice_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
SIGNAL_0,
ADD_LOCAL_IP_ADDRESS_SIGNAL,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_MIN_RTP_PORT,
PROP_MAX_RTP_PORT,
};
static guint gst_webrtc_ice_signals[LAST_SIGNAL] = { 0 };
#define gst_webrtc_ice_parent_class parent_class
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICE, gst_webrtc_ice,
GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_debug,
"webrtcice", 0, "webrtcice"););
/**
* gst_webrtc_ice_add_stream:
* @ice: The #GstWebRTCICE
* @session_id: The session id
*
* Returns: (transfer full) (nullable): The #GstWebRTCICEStream, or %NULL
* Since: 1.22
*/
GstWebRTCICEStream *
gst_webrtc_ice_add_stream (GstWebRTCICE * ice, guint session_id)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream);
return GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream (ice, session_id);
}
/**
* gst_webrtc_ice_find_transport:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @component: The #GstWebRTCICEComponent
*
* Returns: (transfer full) (nullable): The #GstWebRTCICETransport, or %NULL
* Since: 1.22
*/
GstWebRTCICETransport *
gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
GstWebRTCICEStream * stream, GstWebRTCICEComponent component)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport);
return GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport (ice, stream,
component);
}
/**
* gst_webrtc_ice_add_candidate:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @candidate: The ICE candidate
* Since: 1.22
*/
void
gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
GstWebRTCICEStream * stream, const gchar * candidate)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate);
GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate (ice, stream, candidate);
}
/**
* gst_webrtc_ice_set_remote_credentials:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @ufrag: ICE username
* @pwd: ICE password
* Returns: FALSE on error, TRUE otherwise
* Since: 1.22
*/
gboolean
gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials);
return GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials (ice, stream,
ufrag, pwd);
}
/**
* gst_webrtc_ice_add_turn_server:
* @ice: The #GstWebRTCICE
* @uri: URI of the TURN server
* Returns: FALSE on error, TRUE otherwise
* Since: 1.22
*/
gboolean
gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice, const gchar * uri)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server);
return GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server (ice, uri);
}
/**
* gst_webrtc_ice_set_local_credentials:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @ufrag: ICE username
* @pwd: ICE password
* Returns: FALSE on error, TRUE otherwise
* Since: 1.22
*/
gboolean
gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials);
return GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials (ice, stream,
ufrag, pwd);
}
/**
* gst_webrtc_ice_gather_candidates:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* Returns: FALSE on error, TRUE otherwise
* Since: 1.22
*/
gboolean
gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates);
return GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates (ice, stream);
}
/**
* gst_webrtc_ice_set_is_controller:
* @ice: The #GstWebRTCICE
* @controller: TRUE to set as controller
* Since: 1.22
*/
void
gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice, gboolean controller)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller (ice, controller);
}
/**
* gst_webrtc_ice_get_is_controller:
* @ice: The #GstWebRTCICE
* Returns: TRUE if set as controller, FALSE otherwise
* Since: 1.22
*/
gboolean
gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller (ice);
}
/**
* gst_webrtc_ice_set_force_relay:
* @ice: The #GstWebRTCICE
* @force_relay: TRUE to enable force relay
* Since: 1.22
*/
void
gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice, gboolean force_relay)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay (ice, force_relay);
}
/**
* gst_webrtc_ice_set_tos:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @tos: ToS to be set
* Since: 1.22
*/
void
gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream,
guint tos)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos (ice, stream, tos);
}
/**
* gst_webrtc_ice_get_local_candidates:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* Returns: (transfer full)(array zero-terminated=1): List of local candidates
* Since: 1.22
*/
GstWebRTCICECandidateStats *
gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates (ice, stream);
}
/**
* gst_webrtc_ice_get_remote_candidates:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* Returns: (transfer full) (array zero-terminated=1): List of remote candidates
* Since: 1.22
*/
GstWebRTCICECandidateStats *
gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
GstWebRTCICEStream * stream)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates (ice, stream);
}
/**
* gst_webrtc_ice_get_selected_pair:
* @ice: The #GstWebRTCICE
* @stream: The #GstWebRTCICEStream
* @local_stats: (out) (transfer full): A pointer to #GstWebRTCICECandidateStats for local candidate
* @remote_stats: (out) (transfer full): pointer to #GstWebRTCICECandidateStats for remote candidate
*
* Returns: FALSE on failure, otherwise @local_stats @remote_stats will be set
* Since: 1.22
*/
gboolean
gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats,
GstWebRTCICECandidateStats ** remote_stats)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair (ice, stream,
local_stats, remote_stats);
}
/**
* gst_webrtc_ice_candidate_stats_free:
* @stats: The #GstWebRTCICECandidateStats to be free'd
*
* Helper function to free #GstWebRTCICECandidateStats
* Since: 1.22
*/
void
gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats)
{
if (stats) {
g_free (stats->ipaddr);
g_free (stats->url);
}
g_free (stats);
}
/**
* gst_webrtc_ice_candidate_stats_copy:
* @stats: The #GstWebRTCICE
*
* Returns: (transfer full): A copy of @stats
* Since: 1.22
*/
GstWebRTCICECandidateStats *
gst_webrtc_ice_candidate_stats_copy (GstWebRTCICECandidateStats * stats)
{
GstWebRTCICECandidateStats *copy =
g_malloc (sizeof (GstWebRTCICECandidateStats));
*copy = *stats;
copy->ipaddr = g_strdup (stats->ipaddr);
copy->url = g_strdup (stats->url);
return copy;
}
G_DEFINE_BOXED_TYPE (GstWebRTCICECandidateStats, gst_webrtc_ice_candidate_stats,
(GBoxedCopyFunc) gst_webrtc_ice_candidate_stats_copy,
(GBoxedFreeFunc) gst_webrtc_ice_candidate_stats_free);
/**
* gst_webrtc_ice_set_on_ice_candidate:
* @ice: The #GstWebRTCICE
* @func: The #GstWebRTCICEOnCandidateFunc callback function
* @user_data: User data passed to the callback function
* @notify: a #GDestroyNotify when the candidate is no longer needed
* Since: 1.22
*/
void
gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate (ice, func, user_data,
notify);
}
/**
* gst_webrtc_ice_set_stun_server:
* @ice: The #GstWebRTCICE
* @uri: URI of the STUN server
* Since: 1.22
*/
void
gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice, const gchar * uri_s)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server (ice, uri_s);
}
/**
* gst_webrtc_ice_get_stun_server:
* @ice: The #GstWebRTCICE
* Returns: URI of the STUN sever
* Since: 1.22
*/
gchar *
gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server (ice);
}
/**
* gst_webrtc_ice_set_turn_server:
* @ice: The #GstWebRTCICE
* @uri: URI of the TURN sever
* Since: 1.22
*/
void
gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice, const gchar * uri_s)
{
g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server);
GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server (ice, uri_s);
}
/**
* gst_webrtc_ice_get_turn_server:
* @ice: The #GstWebRTCICE
* Returns: URI of the TURN sever
* Since: 1.22
*/
gchar *
gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice)
{
g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server);
return GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server (ice);
}
static void
gst_webrtc_ice_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCICE *ice = GST_WEBRTC_ICE (object);
switch (prop_id) {
case PROP_MIN_RTP_PORT:
ice->min_rtp_port = g_value_get_uint (value);
if (ice->min_rtp_port > ice->max_rtp_port)
g_warning ("Set min-rtp-port to %u which is larger than"
" max-rtp-port %u", ice->min_rtp_port, ice->max_rtp_port);
break;
case PROP_MAX_RTP_PORT:
ice->max_rtp_port = g_value_get_uint (value);
if (ice->min_rtp_port > ice->max_rtp_port)
g_warning ("Set max-rtp-port to %u which is smaller than"
" min-rtp-port %u", ice->max_rtp_port, ice->min_rtp_port);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_ice_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCICE *ice = GST_WEBRTC_ICE (object);
switch (prop_id) {
case PROP_MIN_RTP_PORT:
g_value_set_uint (value, ice->min_rtp_port);
break;
case PROP_MAX_RTP_PORT:
g_value_set_uint (value, ice->max_rtp_port);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
klass->add_stream = NULL;
klass->find_transport = NULL;
klass->gather_candidates = NULL;
klass->add_candidate = NULL;
klass->set_local_credentials = NULL;
klass->set_remote_credentials = NULL;
klass->add_turn_server = NULL;
klass->set_is_controller = NULL;
klass->get_is_controller = NULL;
klass->set_force_relay = NULL;
klass->set_stun_server = NULL;
klass->get_stun_server = NULL;
klass->set_turn_server = NULL;
klass->get_turn_server = NULL;
klass->set_tos = NULL;
klass->set_on_ice_candidate = NULL;
klass->get_local_candidates = NULL;
klass->get_remote_candidates = NULL;
klass->get_selected_pair = NULL;
gobject_class->get_property = gst_webrtc_ice_get_property;
gobject_class->set_property = gst_webrtc_ice_set_property;
/**
* GstWebRTCICE:min-rtp-port:
*
* Minimum port for local rtp port range.
* min-rtp-port must be <= max-rtp-port
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class,
PROP_MIN_RTP_PORT,
g_param_spec_uint ("min-rtp-port", "ICE RTP candidate min port",
"Minimum port for local rtp port range. "
"min-rtp-port must be <= max-rtp-port",
0, 65535, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCICE:max-rtp-port:
*
* Maximum port for local rtp port range.
* min-rtp-port must be <= max-rtp-port
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class,
PROP_MAX_RTP_PORT,
g_param_spec_uint ("max-rtp-port", "ICE RTP candidate max port",
"Maximum port for local rtp port range. "
"max-rtp-port must be >= min-rtp-port",
0, 65535, 65535,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCICE::add-local-ip-address:
* @object: the #GstWebRTCICE
* @address: The local IP address
*
* Add a local IP address to use for ICE candidate gathering. If none
* are supplied, they will be discovered automatically. Calling this signal
* stops automatic ICE gathering.
*
* Returns: whether the address could be added.
*/
gst_webrtc_ice_signals[ADD_LOCAL_IP_ADDRESS_SIGNAL] =
g_signal_new_class_handler ("add-local-ip-address",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
NULL, NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_STRING);
}
static void
gst_webrtc_ice_init (GstWebRTCICE * ice)
{
}