mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-29 21:21:12 +00:00
48ae40f477
According to https://w3c.github.io/webrtc-pc/#set-the-session-description (steps in 4.6.10.), we should be creating and associating transceivers when setting session descriptions. Before this commit, webrtcbin deviated from the spec: 1. Transceivers from sink pads where created when the sink pad was requested, but not associated after setting local description, only when signaling is STABLE. 2. Transceivers from remote offers were not created after applying the the remote description, only when the answer is created, and were then only associated once signaling is STABLE. This commit makes webrtcbin follow the spec more closely with regards to timing of transceivers creation and association. A unit test is added, checking that the transceivers are created and associated after every session description is set. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7156>
121 lines
6.2 KiB
C
121 lines
6.2 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 (GstWebRTCSignalingState state,
|
|
SDPSource source,
|
|
GstWebRTCSessionDescription * sdp,
|
|
GError ** error);
|
|
|
|
G_GNUC_INTERNAL
|
|
GstWebRTCRTPTransceiverDirection _get_direction_from_media (const GstSDPMedia * media);
|
|
G_GNUC_INTERNAL
|
|
GstWebRTCKind _get_kind_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 _get_dtls_setup_from_session (const GstSDPMessage * sdp);
|
|
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);
|
|
|
|
G_GNUC_INTERNAL
|
|
int _get_sctp_port_from_media (const GstSDPMedia * media);
|
|
G_GNUC_INTERNAL
|
|
guint64 _get_sctp_max_message_size_from_media (const GstSDPMedia * media);
|
|
|
|
G_GNUC_INTERNAL
|
|
void _get_ice_credentials_from_sdp_media (const GstSDPMessage * sdp,
|
|
guint media_idx,
|
|
gchar ** ufrag,
|
|
gchar ** pwd);
|
|
G_GNUC_INTERNAL
|
|
gboolean _message_media_is_datachannel (const GstSDPMessage * msg,
|
|
guint media_id);
|
|
G_GNUC_INTERNAL
|
|
guint _message_get_datachannel_index (const GstSDPMessage * msg);
|
|
G_GNUC_INTERNAL
|
|
gboolean _message_has_attribute_key (const GstSDPMessage * msg,
|
|
const gchar * key);
|
|
|
|
G_GNUC_INTERNAL
|
|
gboolean _get_bundle_index (GstSDPMessage * sdp,
|
|
GStrv bundled,
|
|
guint * idx);
|
|
G_GNUC_INTERNAL
|
|
gboolean _parse_bundle (GstSDPMessage * sdp,
|
|
GStrv * bundled,
|
|
GError ** error);
|
|
|
|
G_GNUC_INTERNAL
|
|
const gchar * _media_get_ice_pwd (const GstSDPMessage * msg,
|
|
guint media_idx);
|
|
G_GNUC_INTERNAL
|
|
const gchar * _media_get_ice_ufrag (const GstSDPMessage * msg,
|
|
guint media_idx);
|
|
|
|
G_GNUC_INTERNAL
|
|
gboolean _media_is_bundle_only (const GstSDPMedia * sdp);
|
|
|
|
#endif /* __WEBRTC_UTILS_H__ */
|