mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4b775228bf
Fixes for basic rollback (from have-local-offer or have-remote-offer to stable). Allow having no SDP attached to the webrtc session description in that case, and avoid all the transceiver and ICE update logic normally applied when entering the stable signalling state Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7304>
127 lines
3.3 KiB
C
127 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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if (src->sdp != NULL) {
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gst_sdp_message_copy (src->sdp, &ret->sdp);
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}
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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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if (desc->sdp != NULL) {
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gst_sdp_message_free (desc->sdp);
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}
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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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