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85c75bb23b
This is the equivalent of iceTransportPolicy in the RTCConfiguration dictionary. Only two values are implemented: * all: default behaviour * relay: only gather relay candidates The third member of the iceTransportPolicy enum, "public", is obsolete.
156 lines
5 KiB
C
156 lines
5 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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#ifndef __GST_WEBRTC_BIN_H__
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#define __GST_WEBRTC_BIN_H__
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#include <gst/sdp/sdp.h>
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#include "fwd.h"
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#include "gstwebrtcice.h"
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#include "transportstream.h"
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G_BEGIN_DECLS
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GType gst_webrtc_bin_pad_get_type(void);
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#define GST_TYPE_WEBRTC_BIN_PAD (gst_webrtc_bin_pad_get_type())
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#define GST_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
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#define GST_IS_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
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#define GST_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
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#define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
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#define GST_WEBRTC_BIN_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
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typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
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typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;
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struct _GstWebRTCBinPad
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{
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GstGhostPad parent;
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guint mlineindex;
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GstWebRTCRTPTransceiver *trans;
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gulong block_id;
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GstCaps *received_caps;
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};
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struct _GstWebRTCBinPadClass
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{
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GstGhostPadClass parent_class;
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};
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GType gst_webrtc_bin_get_type(void);
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#define GST_TYPE_WEBRTC_BIN (gst_webrtc_bin_get_type())
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#define GST_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
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#define GST_IS_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
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#define GST_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
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#define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
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#define GST_WEBRTC_BIN_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
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struct _GstWebRTCBin
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{
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GstBin parent;
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GstElement *rtpbin;
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GstElement *rtpfunnel;
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GstWebRTCSignalingState signaling_state;
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GstWebRTCICEGatheringState ice_gathering_state;
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GstWebRTCICEConnectionState ice_connection_state;
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GstWebRTCPeerConnectionState peer_connection_state;
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GstWebRTCSessionDescription *current_local_description;
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GstWebRTCSessionDescription *pending_local_description;
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GstWebRTCSessionDescription *current_remote_description;
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GstWebRTCSessionDescription *pending_remote_description;
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GstWebRTCBundlePolicy bundle_policy;
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GstWebRTCICETransportPolicy ice_transport_policy;
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GstWebRTCBinPrivate *priv;
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};
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struct _GstWebRTCBinClass
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{
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GstBinClass parent_class;
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};
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struct _GstWebRTCBinPrivate
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{
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guint max_sink_pad_serial;
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gboolean bundle;
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GArray *transceivers;
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GArray *session_mid_map;
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GArray *transports;
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GArray *data_channels;
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/* list of data channels we've received a sctp stream for but no data
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* channel protocol for */
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GArray *pending_data_channels;
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GstWebRTCSCTPTransport *sctp_transport;
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TransportStream *data_channel_transport;
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GstWebRTCICE *ice;
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GArray *ice_stream_map;
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GArray *pending_ice_candidates;
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/* peerconnection variables */
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gboolean is_closed;
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gboolean need_negotiation;
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/* peerconnection helper thread for promises */
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GMainContext *main_context;
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GMainLoop *loop;
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GThread *thread;
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GMutex pc_lock;
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GCond pc_cond;
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gboolean running;
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gboolean async_pending;
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GList *pending_pads;
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GList *pending_sink_transceivers;
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/* count of the number of media streams we've offered for uniqueness */
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/* FIXME: overflow? */
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guint media_counter;
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GstStructure *stats;
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};
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typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
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typedef struct
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{
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GstWebRTCBin *webrtc;
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GstWebRTCBinFunc op;
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gpointer data;
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GDestroyNotify notify;
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// GstPromise *promise; /* FIXME */
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} GstWebRTCBinTask;
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void gst_webrtc_bin_enqueue_task (GstWebRTCBin * pc,
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GstWebRTCBinFunc func,
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gpointer data,
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GDestroyNotify notify);
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G_END_DECLS
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#endif /* __GST_WEBRTC_BIN_H__ */
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