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123 lines
3.3 KiB
C
123 lines
3.3 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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/**
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* SECTION:gstwebrtc-sessiondescription
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* @short_description: RTCSessionDescription object
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* @title: GstWebRTCSessionDescription
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*
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* <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "rtcsessiondescription.h"
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#define GST_CAT_DEFAULT gst_webrtc_peerconnection_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/**
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* gst_webrtc_sdp_type_to_string:
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* @type: a #GstWebRTCSDPType
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*
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* Returns: the string representation of @type or "unknown" when @type is not
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* recognized.
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*/
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const gchar *
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gst_webrtc_sdp_type_to_string (GstWebRTCSDPType type)
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{
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switch (type) {
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case GST_WEBRTC_SDP_TYPE_OFFER:
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return "offer";
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case GST_WEBRTC_SDP_TYPE_PRANSWER:
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return "pranswer";
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case GST_WEBRTC_SDP_TYPE_ANSWER:
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return "answer";
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case GST_WEBRTC_SDP_TYPE_ROLLBACK:
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return "rollback";
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default:
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return "unknown";
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}
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}
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/**
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* gst_webrtc_session_description_copy:
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* @src: (transfer none): a #GstWebRTCSessionDescription
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*
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* Returns: (transfer full): a new copy of @src
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*/
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GstWebRTCSessionDescription *
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gst_webrtc_session_description_copy (const GstWebRTCSessionDescription * src)
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{
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GstWebRTCSessionDescription *ret;
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if (!src)
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return NULL;
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ret = g_new0 (GstWebRTCSessionDescription, 1);
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ret->type = src->type;
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gst_sdp_message_copy (src->sdp, &ret->sdp);
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return ret;
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}
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/**
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* gst_webrtc_session_description_free:
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* @desc: (transfer full): a #GstWebRTCSessionDescription
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*
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* Free @desc and all associated resources
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*/
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void
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gst_webrtc_session_description_free (GstWebRTCSessionDescription * desc)
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{
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g_return_if_fail (desc != NULL);
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gst_sdp_message_free (desc->sdp);
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g_free (desc);
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}
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/**
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* gst_webrtc_session_description_new:
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* @type: a #GstWebRTCSDPType
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* @sdp: (transfer full): a #GstSDPMessage
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*
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* Returns: (transfer full): a new #GstWebRTCSessionDescription from @type
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* and @sdp
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*/
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GstWebRTCSessionDescription *
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gst_webrtc_session_description_new (GstWebRTCSDPType type, GstSDPMessage * sdp)
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{
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GstWebRTCSessionDescription *ret;
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ret = g_new0 (GstWebRTCSessionDescription, 1);
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ret->type = type;
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ret->sdp = sdp;
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return ret;
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}
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G_DEFINE_BOXED_TYPE_WITH_CODE (GstWebRTCSessionDescription,
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gst_webrtc_session_description, gst_webrtc_session_description_copy,
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gst_webrtc_session_description_free,
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_peerconnection_debug,
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"webrtcsessiondescription", 0, "webrtcsessiondescription"));
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