gstreamer/ext/webrtc/utils.h
Matthew Waters e2d88f0569 webrtc: propagate more errors through the promise
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
2020-09-14 04:04:29 +00:00

87 lines
3.4 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,
GST_WEBRTC_BIN_ERROR_CLOSED,
GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
} 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__ */