mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 10:41:04 +00:00
1894293d63
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
83 lines
3.8 KiB
C
83 lines
3.8 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;
|
|
|
|
GstWebRTCICEPrivate *priv;
|
|
};
|
|
|
|
struct _GstWebRTCICEClass
|
|
{
|
|
GstObjectClass parent_class;
|
|
};
|
|
|
|
GstWebRTCICE * gst_webrtc_ice_new (void);
|
|
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);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_ICE_H__ */
|