/usr/share/vala-0.40/vapi/gstreamer-webrtc-1.0.vapi is in valac-0.40-vapi 0.40.4-1.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
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 | /* 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, Gst.SDP.Message sdp);
public Gst.WebRTCSessionDescription copy ();
public void free ();
}
[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_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_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_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_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);
}
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