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a100f36b69
Some were leaked when debugging was enabled. Instead just directly use the static strings as-is. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3347>
76 lines
3.2 KiB
C
76 lines
3.2 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __WEBRTC_UTILS_H__
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#define __WEBRTC_UTILS_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc.h>
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#include "fwd.h"
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G_BEGIN_DECLS
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GstPadTemplate * _find_pad_template (GstElement * element,
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GstPadDirection direction,
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GstPadPresence presence,
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const gchar * name);
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GstSDPMessage * _get_latest_sdp (GstWebRTCBin * webrtc);
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GstSDPMessage * _get_latest_offer (GstWebRTCBin * webrtc);
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GstSDPMessage * _get_latest_answer (GstWebRTCBin * webrtc);
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GstSDPMessage * _get_latest_self_generated_sdp (GstWebRTCBin * webrtc);
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GstWebRTCICEStream * _find_ice_stream_for_session (GstWebRTCBin * webrtc,
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guint session_id);
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void _add_ice_stream_item (GstWebRTCBin * webrtc,
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guint session_id,
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GstWebRTCICEStream * stream);
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struct pad_block
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{
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GstElement *element;
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GstPad *pad;
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gulong block_id;
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gpointer user_data;
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GDestroyNotify notify;
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};
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void _free_pad_block (struct pad_block *block);
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struct pad_block * _create_pad_block (GstElement * element,
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GstPad * pad,
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gulong block_id,
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gpointer user_data,
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GDestroyNotify notify);
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G_GNUC_INTERNAL
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const gchar * _enum_value_to_string (GType type, guint value);
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G_GNUC_INTERNAL
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const gchar * _g_checksum_to_webrtc_string (GChecksumType type);
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G_GNUC_INTERNAL
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GstCaps * _rtp_caps_from_media (const GstSDPMedia * media);
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G_GNUC_INTERNAL
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GstWebRTCKind webrtc_kind_from_caps (const GstCaps * caps);
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#define gst_webrtc_kind_to_string(kind) _enum_value_to_string(GST_TYPE_WEBRTC_KIND, kind)
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#define gst_webrtc_rtp_transceiver_direction_to_string(dir) _enum_value_to_string(GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, dir)
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G_END_DECLS
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#endif /* __WEBRTC_UTILS_H__ */
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