vapi/gstreamer-webrtc-1.0.vapi

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