/* 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. */ /** * SECTION:gstwebrtc-sessiondescription * @short_description: RTCSessionDescription object * @title: GstWebRTCSessionDescription * * <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class> */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "rtcsessiondescription.h" #define GST_CAT_DEFAULT gst_webrtc_peerconnection_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); /** * gst_webrtc_sdp_type_to_string: * @type: a #GstWebRTCSDPType * * Returns: the string representation of @type or "unknown" when @type is not * recognized. */ const gchar * gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type) { switch (type) { case GST_WEBRTC_SDP_TYPE_OFFER: return "offer"; case GST_WEBRTC_SDP_TYPE_PRANSWER: return "pranswer"; case GST_WEBRTC_SDP_TYPE_ANSWER: return "answer"; case GST_WEBRTC_SDP_TYPE_ROLLBACK: return "rollback"; default: return "unknown"; } } /** * gst_webrtc_session_description_copy: * @src: (transfer none): a #GstWebRTCSessionDescription * * Returns: (transfer full): a new copy of @src */ GstWebRTCSessionDescription * gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src) { GstWebRTCSessionDescription *ret; if (!src) return NULL; ret = g_new0 (GstWebRTCSessionDescription, 1); ret->type = src->type; gst_sdp_message_copy (src->sdp, &ret->sdp); return ret; } /** * gst_webrtc_session_description_free: * @desc: (transfer full): a #GstWebRTCSessionDescription * * Free @desc and all associated resources */ void gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc) { g_return_if_fail (desc != NULL); gst_sdp_message_free (desc->sdp); g_free (desc); } /** * gst_webrtc_session_description_new: * @type: a #GstWebRTCSDPType * @sdp: (transfer full): a #GstSDPMessage * * Returns: (transfer full): a new #GstWebRTCSessionDescription from @type * and @sdp */ GstWebRTCSessionDescription * gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage * sdp) { GstWebRTCSessionDescription *ret; ret = g_new0 (GstWebRTCSessionDescription, 1); ret->type = type; ret->sdp = sdp; return ret; } G_DEFINE_BOXED_TYPE_WITH_CODE (GstWebRTCSessionDescription, gst_webrtc_session_description, gst_webrtc_session_description_copy, gst_webrtc_session_description_free, GST_DEBUG_CATEGORY_INIT (gst_webrtc_peerconnection_debug, "webrtcsessiondescription", 0, "webrtcsessiondescription"));