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/* 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;
[CCode (has_construct_function = false)]
public WebRTCDTLSTransport (uint session_id);
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 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_data_channel", type_id = "gst_webrtc_data_channel_get_type ()")]
[Version (since = "1.18")]
public abstract class WebRTCDataChannel : GLib.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public GLib.Mutex @lock;
[CCode (has_construct_function = false)]
protected WebRTCDataChannel ();
[NoAccessorMethod]
public uint64 buffered_amount { get; }
[NoAccessorMethod]
public uint64 buffered_amount_low_threshold { get; set; }
[NoAccessorMethod]
public int id { get; construct; }
[NoAccessorMethod]
public string label { owned get; construct; }
[NoAccessorMethod]
public int max_packet_lifetime { get; construct; }
[NoAccessorMethod]
public int max_retransmits { get; construct; }
[NoAccessorMethod]
public bool negotiated { get; construct; }
[NoAccessorMethod]
public bool ordered { get; construct; }
[NoAccessorMethod]
public Gst.WebRTCPriorityType priority { get; construct; }
[NoAccessorMethod]
public string protocol { owned get; construct; }
[NoAccessorMethod]
public Gst.WebRTCDataChannelState ready_state { get; }
[HasEmitter]
public virtual signal void close ();
[HasEmitter]
public signal void on_buffered_amount_low ();
[HasEmitter]
public signal void on_close ();
[HasEmitter]
public signal void on_error (GLib.Error error);
[HasEmitter]
public signal void on_message_data (GLib.Bytes? data);
[HasEmitter]
public signal void on_message_string (string? str);
[HasEmitter]
public signal void on_open ();
[HasEmitter]
public virtual signal void send_data (GLib.Bytes? data);
[HasEmitter]
public virtual signal void send_string (string? str);
}
[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 ()")]
[Version (since = "1.16")]
public class WebRTCRTPReceiver : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak Gst.WebRTCDTLSTransport transport;
[CCode (has_construct_function = false)]
public WebRTCRTPReceiver ();
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_sender", type_id = "gst_webrtc_rtp_sender_get_type ()")]
[Version (since = "1.16")]
public class WebRTCRTPSender : Gst.Object {
[CCode (array_length = false)]
public weak void* _padding[4];
public weak GLib.Array<void*> send_encodings;
public weak Gst.WebRTCDTLSTransport transport;
[CCode (has_construct_function = false)]
public WebRTCRTPSender ();
[Version (since = "1.20")]
public void set_priority (Gst.WebRTCPriorityType priority);
[NoAccessorMethod]
[Version (since = "1.20")]
public Gst.WebRTCPriorityType priority { get; set; }
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_transceiver", type_id = "gst_webrtc_rtp_transceiver_get_type ()")]
[Version (since = "1.16")]
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;
[Version (since = "1.20")]
public Gst.WebRTCKind kind;
public weak string mid;
public uint mline;
public bool stopped;
[CCode (has_construct_function = false)]
protected WebRTCRTPTransceiver ();
[NoAccessorMethod]
[Version (since = "1.18")]
public Gst.WebRTCRTPTransceiverDirection direction { get; set; }
[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 ();
[DestroysInstance]
public void free ();
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_BUNDLE_POLICY_", type_id = "gst_webrtc_bundle_policy_get_type ()")]
[Version (since = "1.16")]
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 ()")]
[Version (since = "1.16")]
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 ()")]
[Version (since = "1.14.1")]
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 ()")]
[Version (since = "1.16")]
public enum WebRTCICETransportPolicy {
ALL,
RELAY
}
[CCode (cheader_filename = "gst/webrtc/webrtc.h", cprefix = "GST_WEBRTC_KIND_", type_id = "gst_webrtc_kind_get_type ()")]
[Version (since = "1.20")]
public enum WebRTCKind {
UNKNOWN,
AUDIO,
VIDEO
}
[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 ()")]
[Version (since = "1.16")]
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 ()")]
[Version (since = "1.16")]
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);
}
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