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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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06893b8b5e
* Add fec / red encoders as direct children of webrtcbin, instead of providing them to rtpbin through the request-fec-encoder signal. That is because they need to be placed before the rtpfunnel, which is placed upstream of rtpbin. * Update configuration of red decoders to set a list of RED payloads on them, instead of setting the pt property. That is because there may be one RED pt per media in the same session. * Connect to request-fec-decoder-full instead of request-fec-decoder, in order to instantiate FEC decoders according to the payload type of the stream. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
90 lines
3.5 KiB
C
90 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 __TRANSPORT_STREAM_H__
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#define __TRANSPORT_STREAM_H__
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#include "fwd.h"
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#include <gst/webrtc/rtptransceiver.h>
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G_BEGIN_DECLS
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GType transport_stream_get_type(void);
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#define GST_TYPE_WEBRTC_TRANSPORT_STREAM (transport_stream_get_type())
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#define TRANSPORT_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStream))
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#define TRANSPORT_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
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#define TRANSPORT_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_TRANSPORT_STREAM,TransportStreamClass))
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typedef struct
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{
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guint8 pt;
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guint media_idx;
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GstCaps *caps;
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} PtMapItem;
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typedef struct
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{
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guint32 ssrc;
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guint media_idx;
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GWeakRef rtpjitterbuffer; /* for stats */
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} SsrcMapItem;
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SsrcMapItem * ssrcmap_item_new (guint32 ssrc,
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guint media_idx);
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struct _TransportStream
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{
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GstObject parent;
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guint session_id; /* session_id */
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gboolean dtls_client;
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gboolean active; /* TRUE if any mline in the bundle/transport is active */
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TransportSendBin *send_bin; /* bin containing all the sending transport elements */
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TransportReceiveBin *receive_bin; /* bin containing all the receiving transport elements */
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GstWebRTCICEStream *stream;
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GstWebRTCDTLSTransport *transport;
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GArray *ptmap; /* array of PtMapItem's */
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GPtrArray *remote_ssrcmap; /* array of SsrcMapItem's */
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gboolean output_connected; /* whether receive bin is connected to rtpbin */
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GstElement *rtxsend;
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GstElement *rtxreceive;
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};
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struct _TransportStreamClass
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{
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GstObjectClass parent_class;
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};
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TransportStream * transport_stream_new (GstWebRTCBin * webrtc,
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guint session_id);
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int transport_stream_get_pt (TransportStream * stream,
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const gchar * encoding_name,
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guint media_idx);
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int * transport_stream_get_all_pt (TransportStream * stream,
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const gchar * encoding_name,
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gsize * pt_len);
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GstCaps * transport_stream_get_caps_for_pt (TransportStream * stream,
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guint pt);
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G_END_DECLS
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#endif /* __TRANSPORT_STREAM_H__ */
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