295 lines
11 KiB
Vala
295 lines
11 KiB
Vala
/* gstreamer-webrtc-1.0.vapi generated by vapigen, do not modify. */
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[CCode (cprefix = "Gst", gir_namespace = "GstWebRTC", gir_version = "1.0", lower_case_cprefix = "gst_")]
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namespace Gst {
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_dtls_transport", type_id = "gst_webrtc_dtls_transport_get_type ()")]
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public class WebRTCDTLSTransport : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCDTLSTransport ();
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[NoAccessorMethod]
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public string certificate { owned get; set; }
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[NoAccessorMethod]
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public bool client { get; set; }
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[NoAccessorMethod]
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public string remote_certificate { owned get; }
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[NoAccessorMethod]
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public uint session_id { get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCDTLSTransportState state { get; }
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[NoAccessorMethod]
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public Gst.WebRTCICETransport transport { owned get; }
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_data_channel", type_id = "gst_webrtc_data_channel_get_type ()")]
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[Version (since = "1.18")]
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public abstract class WebRTCDataChannel : GLib.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCDataChannel ();
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[NoAccessorMethod]
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public uint64 buffered_amount { get; }
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[NoAccessorMethod]
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public uint64 buffered_amount_low_threshold { get; set; }
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[NoAccessorMethod]
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public int id { get; construct; }
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[NoAccessorMethod]
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public string label { owned get; construct; }
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[NoAccessorMethod]
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public int max_packet_lifetime { get; construct; }
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[NoAccessorMethod]
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public int max_retransmits { get; construct; }
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[NoAccessorMethod]
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public bool negotiated { get; construct; }
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[NoAccessorMethod]
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public bool ordered { get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCPriorityType priority { get; construct; }
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[NoAccessorMethod]
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public string protocol { owned get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCDataChannelState ready_state { get; }
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[HasEmitter]
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public signal void close ();
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public signal void on_buffered_amount_low ();
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public signal void on_close ();
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public signal void on_error (GLib.Error error);
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public signal void on_message_data (GLib.Bytes? data);
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public signal void on_message_string (string? data);
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public signal void on_open ();
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[HasEmitter]
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public signal void send_data (GLib.Bytes? data);
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[HasEmitter]
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public signal void send_string (string? str);
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_ice_transport", type_id = "gst_webrtc_ice_transport_get_type ()")]
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public abstract class WebRTCICETransport : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCICETransport ();
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[NoAccessorMethod]
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public Gst.WebRTCICEComponent component { get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCICEGatheringState gathering_state { get; }
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[NoAccessorMethod]
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public Gst.WebRTCICEConnectionState state { get; }
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public signal void on_new_candidate (string object);
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public signal void on_selected_candidate_pair_change ();
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_receiver", type_id = "gst_webrtc_rtp_receiver_get_type ()")]
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public class WebRTCRTPReceiver : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCRTPReceiver ();
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.WebRTCDTLSTransport transport { owned get; }
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_sender", type_id = "gst_webrtc_rtp_sender_get_type ()")]
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public class WebRTCRTPSender : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCRTPSender ();
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[Version (since = "1.20")]
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public void set_priority (Gst.WebRTCPriorityType priority);
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.WebRTCPriorityType priority { get; set; }
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.WebRTCDTLSTransport transport { owned get; }
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_transceiver", type_id = "gst_webrtc_rtp_transceiver_get_type ()")]
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public abstract class WebRTCRTPTransceiver : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCRTPTransceiver ();
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.Caps codec_preferences { owned get; set; }
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.WebRTCRTPTransceiverDirection current_direction { get; }
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[NoAccessorMethod]
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[Version (since = "1.18")]
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public Gst.WebRTCRTPTransceiverDirection direction { get; set; }
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public Gst.WebRTCKind kind { get; }
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[NoAccessorMethod]
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[Version (since = "1.20")]
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public string mid { owned get; }
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[NoAccessorMethod]
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public uint mlineindex { get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCRTPReceiver receiver { owned get; construct; }
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[NoAccessorMethod]
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public Gst.WebRTCRTPSender sender { owned get; construct; }
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_sctp_transport", type_id = "gst_webrtc_sctp_transport_get_type ()")]
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public abstract class WebRTCSCTPTransport : Gst.Object {
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[CCode (has_construct_function = false)]
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protected WebRTCSCTPTransport ();
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[NoAccessorMethod]
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public uint max_channels { get; }
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[NoAccessorMethod]
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public uint64 max_message_size { get; }
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[NoAccessorMethod]
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public Gst.WebRTCSCTPTransportState state { get; }
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[NoAccessorMethod]
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public Gst.WebRTCDTLSTransport transport { owned get; }
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}
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[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 ()")]
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[Compact]
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public class WebRTCSessionDescription {
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public weak Gst.SDP.Message sdp;
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public Gst.WebRTCSDPType type;
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[CCode (has_construct_function = false)]
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public WebRTCSessionDescription (Gst.WebRTCSDPType type, owned Gst.SDP.Message sdp);
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public Gst.WebRTCSessionDescription copy ();
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[DestroysInstance]
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public void free ();
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_BUNDLE_POLICY_", type_id = "gst_webrtc_bundle_policy_get_type ()")]
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[Version (since = "1.16")]
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public enum WebRTCBundlePolicy {
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NONE,
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BALANCED,
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MAX_COMPAT,
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MAX_BUNDLE
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_SETUP_", type_id = "gst_webrtc_dtls_setup_get_type ()")]
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public enum WebRTCDTLSSetup {
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NONE,
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ACTPASS,
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ACTIVE,
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PASSIVE
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DTLS_TRANSPORT_STATE_", type_id = "gst_webrtc_dtls_transport_state_get_type ()")]
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public enum WebRTCDTLSTransportState {
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NEW,
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CLOSED,
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FAILED,
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CONNECTING,
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CONNECTED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_DATA_CHANNEL_STATE_", type_id = "gst_webrtc_data_channel_state_get_type ()")]
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[Version (since = "1.16")]
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public enum WebRTCDataChannelState {
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NEW,
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CONNECTING,
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OPEN,
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CLOSING,
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CLOSED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_FEC_TYPE_", type_id = "gst_webrtc_fec_type_get_type ()")]
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[Version (since = "1.14.1")]
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public enum WebRTCFECType {
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NONE,
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ULP_RED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_COMPONENT_", type_id = "gst_webrtc_ice_component_get_type ()")]
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public enum WebRTCICEComponent {
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RTP,
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RTCP
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_CONNECTION_STATE_", type_id = "gst_webrtc_ice_connection_state_get_type ()")]
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public enum WebRTCICEConnectionState {
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NEW,
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CHECKING,
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CONNECTED,
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COMPLETED,
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FAILED,
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DISCONNECTED,
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CLOSED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_GATHERING_STATE_", type_id = "gst_webrtc_ice_gathering_state_get_type ()")]
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public enum WebRTCICEGatheringState {
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NEW,
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GATHERING,
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COMPLETE
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_ROLE_", type_id = "gst_webrtc_ice_role_get_type ()")]
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public enum WebRTCICERole {
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CONTROLLED,
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CONTROLLING
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_ICE_TRANSPORT_POLICY_", type_id = "gst_webrtc_ice_transport_policy_get_type ()")]
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[Version (since = "1.16")]
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public enum WebRTCICETransportPolicy {
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ALL,
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RELAY
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_KIND_", type_id = "gst_webrtc_kind_get_type ()")]
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[Version (since = "1.20")]
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public enum WebRTCKind {
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UNKNOWN,
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AUDIO,
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VIDEO
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PEER_CONNECTION_STATE_", type_id = "gst_webrtc_peer_connection_state_get_type ()")]
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public enum WebRTCPeerConnectionState {
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NEW,
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CONNECTING,
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CONNECTED,
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DISCONNECTED,
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FAILED,
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CLOSED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_PRIORITY_TYPE_", type_id = "gst_webrtc_priority_type_get_type ()")]
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[Version (since = "1.16")]
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public enum WebRTCPriorityType {
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VERY_LOW,
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LOW,
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MEDIUM,
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HIGH
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_", type_id = "gst_webrtc_rtp_transceiver_direction_get_type ()")]
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public enum WebRTCRTPTransceiverDirection {
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NONE,
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INACTIVE,
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SENDONLY,
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RECVONLY,
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SENDRECV
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SCTP_TRANSPORT_STATE_", type_id = "gst_webrtc_sctp_transport_state_get_type ()")]
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[Version (since = "1.16")]
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public enum WebRTCSCTPTransportState {
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NEW,
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CONNECTING,
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CONNECTED,
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CLOSED
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SDP_TYPE_", type_id = "gst_webrtc_sdp_type_get_type ()")]
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public enum WebRTCSDPType {
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OFFER,
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PRANSWER,
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ANSWER,
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ROLLBACK;
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[CCode (cname = "gst_webrtc_sdp_type_to_string")]
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public unowned string to_string ();
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_SIGNALING_STATE_", type_id = "gst_webrtc_signaling_state_get_type ()")]
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public enum WebRTCSignalingState {
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STABLE,
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CLOSED,
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HAVE_LOCAL_OFFER,
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HAVE_REMOTE_OFFER,
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HAVE_LOCAL_PRANSWER,
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HAVE_REMOTE_PRANSWER
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_STATS_", type_id = "gst_webrtc_stats_type_get_type ()")]
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public enum WebRTCStatsType {
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CODEC,
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INBOUND_RTP,
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OUTBOUND_RTP,
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REMOTE_INBOUND_RTP,
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REMOTE_OUTBOUND_RTP,
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CSRC,
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PEER_CONNECTION,
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DATA_CHANNEL,
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STREAM,
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TRANSPORT,
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CANDIDATE_PAIR,
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LOCAL_CANDIDATE,
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REMOTE_CANDIDATE,
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CERTIFICATE
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}
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[CCode (cheader_filename = "gst/webrtc/webrtc.h")]
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[Version (replacement = "WebRTCSDPType.to_string")]
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public static unowned string webrtc_sdp_type_to_string (Gst.WebRTCSDPType type);
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}
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