gstreamer/subprojects/gst-plugins-bad/ext/webrtc/utils.h
Jan Schmidt ef71c1319a webrtcbin: Improve SDP intersection for Opus
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>
2024-01-25 13:37:21 +00:00

79 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
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
const gchar * _enum_value_to_string (GType type, guint value);
G_GNUC_INTERNAL
const gchar * _g_checksum_to_webrtc_string (GChecksumType type);
G_GNUC_INTERNAL
void _remove_optional_offer_fields (GstCaps *offer_caps);
G_GNUC_INTERNAL
GstCaps * _rtp_caps_from_media (const GstSDPMedia * media);
G_GNUC_INTERNAL
GstWebRTCKind webrtc_kind_from_caps (const GstCaps * caps);
G_GNUC_INTERNAL
char * _get_msid_from_media (const GstSDPMedia * media);
#define gst_webrtc_kind_to_string(kind) _enum_value_to_string(GST_TYPE_WEBRTC_KIND, kind)
#define gst_webrtc_rtp_transceiver_direction_to_string(dir) _enum_value_to_string(GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, dir)
G_END_DECLS
#endif /* __WEBRTC_UTILS_H__ */