2 * Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 #ifndef __GST_WEBRTC_BIN_H__
21 #define __GST_WEBRTC_BIN_H__
23 #include <gst/sdp/sdp.h>
25 #include "gstwebrtcice.h"
26 #include "transportstream.h"
30 GType gst_webrtc_bin_pad_get_type(void);
31 #define GST_TYPE_WEBRTC_BIN_PAD (gst_webrtc_bin_pad_get_type())
32 #define GST_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPad))
33 #define GST_IS_WEBRTC_BIN_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN_PAD))
34 #define GST_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
35 #define GST_IS_WEBRTC_BIN_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN_PAD))
36 #define GST_WEBRTC_BIN_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN_PAD,GstWebRTCBinPadClass))
38 typedef struct _GstWebRTCBinPad GstWebRTCBinPad;
39 typedef struct _GstWebRTCBinPadClass GstWebRTCBinPadClass;
41 struct _GstWebRTCBinPad
47 GstWebRTCRTPTransceiver *trans;
50 GstCaps *received_caps;
53 struct _GstWebRTCBinPadClass
55 GstGhostPadClass parent_class;
58 GType gst_webrtc_bin_get_type(void);
59 #define GST_TYPE_WEBRTC_BIN (gst_webrtc_bin_get_type())
60 #define GST_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_BIN,GstWebRTCBin))
61 #define GST_IS_WEBRTC_BIN(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_BIN))
62 #define GST_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
63 #define GST_IS_WEBRTC_BIN_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_BIN))
64 #define GST_WEBRTC_BIN_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_BIN,GstWebRTCBinClass))
71 GstElement *rtpfunnel;
73 GstWebRTCSignalingState signaling_state;
74 GstWebRTCICEGatheringState ice_gathering_state;
76 GstWebRTCICEGatheringState pending_ice_gathering_state;
78 GstWebRTCICEConnectionState ice_connection_state;
79 GstWebRTCPeerConnectionState peer_connection_state;
81 GstWebRTCSessionDescription *current_local_description;
82 GstWebRTCSessionDescription *pending_local_description;
83 GstWebRTCSessionDescription *current_remote_description;
84 GstWebRTCSessionDescription *pending_remote_description;
86 GstWebRTCBundlePolicy bundle_policy;
87 GstWebRTCICETransportPolicy ice_transport_policy;
89 GstWebRTCBinPrivate *priv;
92 struct _GstWebRTCBinClass
94 GstBinClass parent_class;
97 struct _GstWebRTCBinPrivate
99 guint max_sink_pad_serial;
102 GPtrArray *transceivers;
103 GArray *session_mid_map;
104 GPtrArray *transports;
105 GPtrArray *data_channels;
106 /* list of data channels we've received a sctp stream for but no data
107 * channel protocol for */
108 GPtrArray *pending_data_channels;
112 GstWebRTCSCTPTransport *sctp_transport;
113 TransportStream *data_channel_transport;
116 GArray *ice_stream_map;
118 GArray *pending_remote_ice_candidates;
119 GArray *pending_local_ice_candidates;
121 /* peerconnection variables */
123 gboolean need_negotiation;
125 /* peerconnection helper thread for promises */
126 GMainContext *main_context;
133 gboolean async_pending;
136 GList *pending_sink_transceivers;
138 /* count of the number of media streams we've offered for uniqueness */
139 /* FIXME: overflow? */
141 /* the number of times create_offer has been called for the version field */
143 GstWebRTCSessionDescription *last_generated_offer;
144 GstWebRTCSessionDescription *last_generated_answer;
146 gboolean tos_attached;
147 #ifdef TIZEN_FEATURE_IMPORT_NETSIM
149 gfloat drop_probability_sender;
153 typedef void (*GstWebRTCBinFunc) (GstWebRTCBin * webrtc, gpointer data);
157 GstWebRTCBin *webrtc;
160 GDestroyNotify notify;
164 gboolean gst_webrtc_bin_enqueue_task (GstWebRTCBin * pc,
165 GstWebRTCBinFunc func,
167 GDestroyNotify notify,
168 GstPromise *promise);
172 #endif /* __GST_WEBRTC_BIN_H__ */