gstreamer/subprojects/gst-plugins-bad/ext/webrtc/webrtctransceiver.h
Mathieu Duponchelle 06fec40f45 webrtcbin: fix msid line and allow customization
From https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-msid-16:

> Multiple media descriptions with the same value for msid-id and
> msid-appdata are not permitted.

Our previous implementation of simply using the CNAME as the msid
identifier and the name of the transceiver as the msid appdata was
misguided and incorrect, and created issues when bundling multiple
video streams together: the ontrack event was emitted with the same
streams for the two bundled medias, at least in Firefox.

Instead, use the transceiver name as the identifier, and expose
a msid-appdata property on transceivers to allow for further
customization by the application. When the property is not set,
msid-appdata can be left empty as it is specified as optional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2003>
2022-03-24 16:43:29 +00:00

77 lines
2.9 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_TRANSCEIVER_H__
#define __WEBRTC_TRANSCEIVER_H__
#include "fwd.h"
#include <gst/webrtc/rtptransceiver.h>
#include "gst/webrtc/webrtc-priv.h"
#include "transportstream.h"
G_BEGIN_DECLS
GType webrtc_transceiver_get_type(void);
#define WEBRTC_TYPE_TRANSCEIVER (webrtc_transceiver_get_type())
#define WEBRTC_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiver))
#define WEBRTC_IS_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),WEBRTC_TYPE_TRANSCEIVER))
#define WEBRTC_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
#define WEBRTC_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,WEBRTC_TYPE_TRANSCEIVER,WebRTCTransceiverClass))
struct _WebRTCTransceiver
{
GstWebRTCRTPTransceiver parent;
TransportStream *stream;
GstStructure *local_rtx_ssrc_map;
guint current_ssrc;
GstEvent *ssrc_event;
/* Properties */
GstWebRTCFECType fec_type;
guint fec_percentage;
gboolean do_nack;
gchar *msid_appdata;
GstCaps *last_configured_caps;
gboolean mline_locked;
GstElement *ulpfecdec;
GstElement *ulpfecenc;
GstElement *redenc;
};
struct _WebRTCTransceiverClass
{
GstWebRTCRTPTransceiverClass parent_class;
};
WebRTCTransceiver * webrtc_transceiver_new (GstWebRTCBin * webrtc,
GstWebRTCRTPSender * sender,
GstWebRTCRTPReceiver * receiver);
void webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
TransportStream * stream);
GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans);
G_END_DECLS
#endif /* __WEBRTC_TRANSCEIVER_H__ */