mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-05 17:09:48 +00:00
cef839533e
Fixes multiple errors when a webrtcbin renegotiation can switch between the offerer and the answerer.
85 lines
3.3 KiB
C
85 lines
3.3 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_UTILS_H__
|
|
#define __WEBRTC_UTILS_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/webrtc/webrtc.h>
|
|
#include "fwd.h"
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
|
|
GQuark gst_webrtc_bin_error_quark (void);
|
|
|
|
typedef enum
|
|
{
|
|
GST_WEBRTC_BIN_ERROR_FAILED,
|
|
GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
|
|
GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
|
|
GST_WEBRTC_BIN_ERROR_INVALID_STATE,
|
|
GST_WEBRTC_BIN_ERROR_BAD_SDP,
|
|
GST_WEBRTC_BIN_ERROR_FINGERPRINT,
|
|
GST_WEBRTC_BIN_ERROR_SCTP_FAILURE,
|
|
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
|
|
} GstWebRTCError;
|
|
|
|
GstPadTemplate * _find_pad_template (GstElement * element,
|
|
GstPadDirection direction,
|
|
GstPadPresence presence,
|
|
const gchar * name);
|
|
|
|
GstSDPMessage * _get_latest_sdp (GstWebRTCBin * webrtc);
|
|
GstSDPMessage * _get_latest_offer (GstWebRTCBin * webrtc);
|
|
GstSDPMessage * _get_latest_answer (GstWebRTCBin * webrtc);
|
|
GstSDPMessage * _get_latest_self_generated_sdp (GstWebRTCBin * webrtc);
|
|
|
|
GstWebRTCICEStream * _find_ice_stream_for_session (GstWebRTCBin * webrtc,
|
|
guint session_id);
|
|
void _add_ice_stream_item (GstWebRTCBin * webrtc,
|
|
guint session_id,
|
|
GstWebRTCICEStream * stream);
|
|
|
|
struct pad_block
|
|
{
|
|
GstElement *element;
|
|
GstPad *pad;
|
|
gulong block_id;
|
|
gpointer user_data;
|
|
GDestroyNotify notify;
|
|
};
|
|
|
|
void _free_pad_block (struct pad_block *block);
|
|
struct pad_block * _create_pad_block (GstElement * element,
|
|
GstPad * pad,
|
|
gulong block_id,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
G_GNUC_INTERNAL
|
|
gchar * _enum_value_to_string (GType type, guint value);
|
|
G_GNUC_INTERNAL
|
|
const gchar * _g_checksum_to_webrtc_string (GChecksumType type);
|
|
G_GNUC_INTERNAL
|
|
GstCaps * _rtp_caps_from_media (const GstSDPMedia * media);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __WEBRTC_UTILS_H__ */
|