1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
|
/* gstreamer-webrtc-1.0.vapi generated by vapigen, do not modify. */
[CCode (cprefix = "Gst", gir_namespace = "GstWebRTC", gir_version = "1.0", lower_case_cprefix = "gst_")]
namespace Gst {
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_dtls_transport", type_id = "gst_webrtc_dtls_transport_get_type ()")]
public class WebRTCDTLSTransport : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak Gst.Element dtlssrtpdec;
public weak Gst.Element dtlssrtpenc;
public bool is_rtcp;
[CCode (has_construct_function = false)]
public WebRTCDTLSTransport (uint session_id, bool rtcp);
public void set_transport (Gst.WebRTCICETransport ice);
[NoAccessorMethod]
public string certificate { owned get; set; }
[NoAccessorMethod]
public bool client { get; set; }
[NoAccessorMethod]
public string remote_certificate { owned get; }
[NoAccessorMethod]
public bool rtcp { get; construct; }
[NoAccessorMethod]
public uint session_id { get; construct; }
[NoAccessorMethod]
public Gst.WebRTCDTLSTransportState state { get; }
[NoAccessorMethod]
public Gst.WebRTCICETransport transport { owned get; }
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_ice_transport", type_id = "gst_webrtc_ice_transport_get_type ()")]
public abstract class WebRTCICETransport : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public Gst.WebRTCICERole role;
public weak Gst.Element sink;
public weak Gst.Element src;
[CCode (has_construct_function = false)]
protected WebRTCICETransport ();
public void connection_state_change (Gst.WebRTCICEConnectionState new_state);
[NoWrapper]
public virtual bool gather_candidates ();
public void gathering_state_change (Gst.WebRTCICEGatheringState new_state);
public void new_candidate (uint stream_id, Gst.WebRTCICEComponent component, string attr);
public void selected_pair_change ();
[NoAccessorMethod]
public Gst.WebRTCICEComponent component { get; construct; }
[NoAccessorMethod]
public Gst.WebRTCICEGatheringState gathering_state { get; }
[NoAccessorMethod]
public Gst.WebRTCICEConnectionState state { get; }
public signal void on_new_candidate (string object);
public signal void on_selected_candidate_pair_change ();
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_receiver", type_id = "gst_webrtc_rtp_receiver_get_type ()")]
public class WebRTCRTPReceiver : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak Gst.WebRTCDTLSTransport rtcp_transport;
public weak Gst.WebRTCDTLSTransport transport;
[CCode (has_construct_function = false)]
public WebRTCRTPReceiver ();
public void set_rtcp_transport (Gst.WebRTCDTLSTransport transport);
public void set_transport (Gst.WebRTCDTLSTransport transport);
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_sender", type_id = "gst_webrtc_rtp_sender_get_type ()")]
public class WebRTCRTPSender : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak Gst.WebRTCDTLSTransport rtcp_transport;
public weak GLib.Array<void*> send_encodings;
public weak Gst.WebRTCDTLSTransport transport;
[CCode (has_construct_function = false)]
public WebRTCRTPSender ();
public void set_rtcp_transport (Gst.WebRTCDTLSTransport transport);
public void set_transport (Gst.WebRTCDTLSTransport transport);
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_transceiver", type_id = "gst_webrtc_rtp_transceiver_get_type ()")]
public abstract class WebRTCRTPTransceiver : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak Gst.Caps codec_preferences;
public Gst.WebRTCRTPTransceiverDirection current_direction;
public Gst.WebRTCRTPTransceiverDirection direction;
public weak string mid;
public uint mline;
public bool stopped;
[CCode (has_construct_function = false)]
protected WebRTCRTPTransceiver ();
[NoAccessorMethod]
public uint mlineindex { get; construct; }
[NoAccessorMethod]
public Gst.WebRTCRTPReceiver receiver { owned get; construct; }
[NoAccessorMethod]
public Gst.WebRTCRTPSender sender { owned get; construct; }
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", copy_function = "g_boxed_copy", free_function = "g_boxed_free", lower_case_csuffix = "webrtc_session_description", type_id = "gst_webrtc_session_description_get_type ()")]
[Compact]
public class WebRTCSessionDescription {
public weak Gst.SDP.Message sdp;
public Gst.WebRTCSDPType type;
[CCode (has_construct_function = false)]
public WebRTCSessionDescription (Gst.WebRTCSDPType type, owned Gst.SDP.Message sdp);
public Gst.WebRTCSessionDescription copy ();
public void free ();
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_BUNDLE_POLICY_", type_id = "gst_webrtc_bundle_policy_get_type ()")]
public enum WebRTCBundlePolicy {
NONE,
BALANCED,
MAX_COMPAT,
MAX_BUNDLE
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_SETUP_", type_id = "gst_webrtc_dtls_setup_get_type ()")]
public enum WebRTCDTLSSetup {
NONE,
ACTPASS,
ACTIVE,
PASSIVE
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_TRANSPORT_STATE_", type_id = "gst_webrtc_dtls_transport_state_get_type ()")]
public enum WebRTCDTLSTransportState {
NEW,
CLOSED,
FAILED,
CONNECTING,
CONNECTED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DATA_CHANNEL_STATE_", type_id = "gst_webrtc_data_channel_state_get_type ()")]
public enum WebRTCDataChannelState {
NEW,
CONNECTING,
OPEN,
CLOSING,
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_FEC_TYPE_", type_id = "gst_webrtc_fec_type_get_type ()")]
public enum WebRTCFECType {
NONE,
ULP_RED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_COMPONENT_", type_id = "gst_webrtc_ice_component_get_type ()")]
public enum WebRTCICEComponent {
RTP,
RTCP
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_CONNECTION_STATE_", type_id = "gst_webrtc_ice_connection_state_get_type ()")]
public enum WebRTCICEConnectionState {
NEW,
CHECKING,
CONNECTED,
COMPLETED,
FAILED,
DISCONNECTED,
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_GATHERING_STATE_", type_id = "gst_webrtc_ice_gathering_state_get_type ()")]
public enum WebRTCICEGatheringState {
NEW,
GATHERING,
COMPLETE
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_ROLE_", type_id = "gst_webrtc_ice_role_get_type ()")]
public enum WebRTCICERole {
CONTROLLED,
CONTROLLING
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_TRANSPORT_POLICY_", type_id = "gst_webrtc_ice_transport_policy_get_type ()")]
public enum WebRTCICETransportPolicy {
ALL,
RELAY
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PEER_CONNECTION_STATE_", type_id = "gst_webrtc_peer_connection_state_get_type ()")]
public enum WebRTCPeerConnectionState {
NEW,
CONNECTING,
CONNECTED,
DISCONNECTED,
FAILED,
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PRIORITY_TYPE_", type_id = "gst_webrtc_priority_type_get_type ()")]
public enum WebRTCPriorityType {
VERY_LOW,
LOW,
MEDIUM,
HIGH
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_", type_id = "gst_webrtc_rtp_transceiver_direction_get_type ()")]
public enum WebRTCRTPTransceiverDirection {
NONE,
INACTIVE,
SENDONLY,
RECVONLY,
SENDRECV
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SCTP_TRANSPORT_STATE_", type_id = "gst_webrtc_sctp_transport_state_get_type ()")]
public enum WebRTCSCTPTransportState {
NEW,
CONNECTING,
CONNECTED,
CLOSED
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SDP_TYPE_", type_id = "gst_webrtc_sdp_type_get_type ()")]
public enum WebRTCSDPType {
OFFER,
PRANSWER,
ANSWER,
ROLLBACK
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SIGNALING_STATE_", type_id = "gst_webrtc_signaling_state_get_type ()")]
public enum WebRTCSignalingState {
STABLE,
CLOSED,
HAVE_LOCAL_OFFER,
HAVE_REMOTE_OFFER,
HAVE_LOCAL_PRANSWER,
HAVE_REMOTE_PRANSWER
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_STATS_", type_id = "gst_webrtc_stats_type_get_type ()")]
public enum WebRTCStatsType {
CODEC,
INBOUND_RTP,
OUTBOUND_RTP,
REMOTE_INBOUND_RTP,
REMOTE_OUTBOUND_RTP,
CSRC,
PEER_CONNECTION,
DATA_CHANNEL,
STREAM,
TRANSPORT,
CANDIDATE_PAIR,
LOCAL_CANDIDATE,
REMOTE_CANDIDATE,
CERTIFICATE
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h")]
public static unowned string webrtc_sdp_type_to_string (Gst.WebRTCSDPType type);
}
|
