diff options
Diffstat (limited to 'gst-libs/gst/webrtc/rtcsessiondescription.h')
-rw-r--r-- | gst-libs/gst/webrtc/rtcsessiondescription.h | 58 |
1 files changed, 58 insertions, 0 deletions
diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.h b/gst-libs/gst/webrtc/rtcsessiondescription.h new file mode 100644 index 000000000..080d21c7e --- /dev/null +++ b/gst-libs/gst/webrtc/rtcsessiondescription.h @@ -0,0 +1,58 @@ +/* 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_EXPORT +const gchar * gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type); + +#define GST_TYPE_WEBRTC_SESSION_DESCRIPTION (gst_webrtc_session_description_get_type()) +GST_EXPORT +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_EXPORT +GstWebRTCSessionDescription * gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage *sdp); +GST_EXPORT +GstWebRTCSessionDescription * gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src); +GST_EXPORT +void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc); + +G_END_DECLS + +#endif /* __GST_WEBRTC_PEERCONNECTION_H__ */ |