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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ef71c1319a
Remove optional sprop-stereo and sprop-maxcapture fields from Opus remote offer caps before intersecting with local codec preferences. According to https://datatracker.ietf.org/doc/html/rfc7587#section-7.1 those fields are sender-only informative, and don't affect interoperability. Fixes cases where the webrtc media will end up receive-only if the local side wants to send stereo but the remote is sending mono, or vice versa. There may be other fields in other codecs, so the implementation anticipates needing to add further fields and codecs in the future. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5993>
80 lines
3.4 KiB
C
80 lines
3.4 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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void _remove_optional_offer_fields (GstCaps *offer_caps);
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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_GNUC_INTERNAL
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char * _get_msid_from_media (const GstSDPMedia * media);
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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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