gstreamer/ext/webrtc/webrtcsdp.h
Matthew Waters 1894293d63 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
2018-02-02 15:02:21 +11:00

80 lines
3.7 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 __WEBRTC_SDP_H__
#define __WEBRTC_SDP_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc.h>
#include "fwd.h"
G_BEGIN_DECLS
typedef enum
{
SDP_NONE,
SDP_LOCAL,
SDP_REMOTE,
} SDPSource;
G_GNUC_INTERNAL
const gchar * _sdp_source_to_string (SDPSource source);
G_GNUC_INTERNAL
gboolean validate_sdp (GstWebRTCBin * webrtc,
SDPSource source,
GstWebRTCSessionDescription * sdp,
GError ** error);
G_GNUC_INTERNAL
GstWebRTCRTPTransceiverDirection _get_direction_from_media (const GstSDPMedia * media);
G_GNUC_INTERNAL
GstWebRTCRTPTransceiverDirection _intersect_answer_directions (GstWebRTCRTPTransceiverDirection offer,
GstWebRTCRTPTransceiverDirection answer);
G_GNUC_INTERNAL
void _media_replace_direction (GstSDPMedia * media,
GstWebRTCRTPTransceiverDirection direction);
G_GNUC_INTERNAL
GstWebRTCRTPTransceiverDirection _get_final_direction (GstWebRTCRTPTransceiverDirection local_dir,
GstWebRTCRTPTransceiverDirection remote_dir);
G_GNUC_INTERNAL
GstWebRTCDTLSSetup _get_dtls_setup_from_media (const GstSDPMedia * media);
G_GNUC_INTERNAL
GstWebRTCDTLSSetup _intersect_dtls_setup (GstWebRTCDTLSSetup offer);
G_GNUC_INTERNAL
void _media_replace_setup (GstSDPMedia * media,
GstWebRTCDTLSSetup setup);
G_GNUC_INTERNAL
GstWebRTCDTLSSetup _get_final_setup (GstWebRTCDTLSSetup local_setup,
GstWebRTCDTLSSetup remote_setup);
G_GNUC_INTERNAL
gchar * _generate_fingerprint_from_certificate (gchar * certificate,
GChecksumType checksum_type);
G_GNUC_INTERNAL
void _generate_ice_credentials (gchar ** ufrag,
gchar ** password);
G_GNUC_INTERNAL
gboolean _media_has_attribute_key (const GstSDPMedia * media,
const gchar * key);
#endif /* __WEBRTC_UTILS_H__ */