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da2bd55177
libnice doesn't touch the kernel buffer sizes. When dealing with RTP data, it's generally advisable to increase them to avoid dropping packets locally. This is especially important when running multiple higher bitrate streams at the same time. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1366>
61 lines
2.4 KiB
C
61 lines
2.4 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 __GST_WEBRTC_NICE_TRANSPORT_H__
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#define __GST_WEBRTC_NICE_TRANSPORT_H__
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#include <gst/gst.h>
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/* libnice */
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#include <agent.h>
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#include <gst/webrtc/webrtc.h>
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#include "gstwebrtcice.h"
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G_BEGIN_DECLS
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GType gst_webrtc_nice_transport_get_type(void);
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#define GST_TYPE_WEBRTC_NICE_TRANSPORT (gst_webrtc_nice_transport_get_type())
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#define GST_WEBRTC_NICE_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_NICE_TRANSPORT,GstWebRTCNiceTransport))
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#define GST_IS_WEBRTC_NICE_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_NICE_TRANSPORT))
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#define GST_WEBRTC_NICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_NICE_TRANSPORT,GstWebRTCNiceTransportClass))
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#define GST_IS_WEBRTC_NICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_NICE_TRANSPORT))
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#define GST_WEBRTC_NICE_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_NICE_TRANSPORT,GstWebRTCNiceTransportClass))
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struct _GstWebRTCNiceTransport
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{
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GstWebRTCICETransport parent;
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GstWebRTCICEStream *stream;
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GstWebRTCNiceTransportPrivate *priv;
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};
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struct _GstWebRTCNiceTransportClass
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{
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GstWebRTCICETransportClass parent_class;
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};
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GstWebRTCNiceTransport * gst_webrtc_nice_transport_new (GstWebRTCICEStream * stream,
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GstWebRTCICEComponent component);
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void gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice);
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G_END_DECLS
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#endif /* __GST_WEBRTC_NICE_TRANSPORT_H__ */
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