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SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
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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* <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
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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: 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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