/usr/include/gstreamer-1.0/gst/webrtc/rtcsessiondescription.h is in libgstreamer-plugins-bad1.0-dev 1.14.0-1ubuntu1.
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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 | /* 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 __GST_WEBRTC_SESSION_DESCRIPTION_H__
#define __GST_WEBRTC_SESSION_DESCRIPTION_H__
#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#include <gst/webrtc/webrtc_fwd.h>
G_BEGIN_DECLS
GST_WEBRTC_API
const gchar * gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type);
#define GST_TYPE_WEBRTC_SESSION_DESCRIPTION (gst_webrtc_session_description_get_type())
GST_WEBRTC_API
GType gst_webrtc_session_description_get_type (void);
/**
* GstWebRTCSessionDescription:
* type: the #GstWebRTCSDPType of the description
* sdp: the #GstSDPMessage of the description
*
* See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
*/
struct _GstWebRTCSessionDescription
{
GstWebRTCSDPType type;
GstSDPMessage *sdp;
};
GST_WEBRTC_API
GstWebRTCSessionDescription * gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage *sdp);
GST_WEBRTC_API
GstWebRTCSessionDescription * gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src);
GST_WEBRTC_API
void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc);
G_END_DECLS
#endif /* __GST_WEBRTC_PEERCONNECTION_H__ */
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