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Now intersect against pads on both sides if they are available. If the intersection fails, we now just reject the creation of the offer or answer as it means that the codec_preferences are too restrictive or that the caps on both sides the webrtcbin are not compatible. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
91 lines
3.5 KiB
C
91 lines
3.5 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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#define GST_WEBRTC_BIN_ERROR gst_webrtc_bin_error_quark ()
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GQuark gst_webrtc_bin_error_quark (void);
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typedef enum
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{
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GST_WEBRTC_BIN_ERROR_FAILED,
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GST_WEBRTC_BIN_ERROR_INVALID_SYNTAX,
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GST_WEBRTC_BIN_ERROR_INVALID_MODIFICATION,
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GST_WEBRTC_BIN_ERROR_INVALID_STATE,
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GST_WEBRTC_BIN_ERROR_BAD_SDP,
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GST_WEBRTC_BIN_ERROR_FINGERPRINT,
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GST_WEBRTC_BIN_ERROR_SCTP_FAILURE,
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GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
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GST_WEBRTC_BIN_ERROR_CLOSED,
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GST_WEBRTC_BIN_ERROR_NOT_IMPLEMENTED,
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GST_WEBRTC_BIN_ERROR_IMPOSSIBLE_MLINE_RESTRICTION,
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GST_WEBRTC_BIN_ERROR_CAPS_NEGOTIATION_FAILED
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} GstWebRTCError;
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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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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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G_END_DECLS
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#endif /* __WEBRTC_UTILS_H__ */
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