/* 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 sealed class WebRTCDTLSTransport : Gst.Object {
		[CCode (has_construct_function = false)]
		protected WebRTCDTLSTransport ();
		[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 (has_construct_function = false)]
		protected WebRTCDataChannel ();
		[Version (since = "1.22")]
		public bool send_data_full (GLib.Bytes? data) throws GLib.Error;
		[Version (since = "1.22")]
		public bool send_string_full (string? str) throws GLib.Error;
		[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 signal void close ();
		public signal void on_buffered_amount_low ();
		public signal void on_close ();
		public signal void on_error (GLib.Error error);
		public signal void on_message_data (GLib.Bytes? data);
		public signal void on_message_string (string? data);
		public signal void on_open ();
		[HasEmitter]
		public signal void send_data (GLib.Bytes? data);
		[HasEmitter]
		public signal void send_string (string? str);
	}
	[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_ice", type_id = "gst_webrtc_ice_get_type ()")]
	[Version (since = "1.22")]
	public abstract class WebRTCICE : Gst.Object {
		[CCode (array_length = false)]
		public weak void* _gst_reserved[4];
		public Gst.WebRTCICEConnectionState ice_connection_state;
		public Gst.WebRTCICEGatheringState ice_gathering_state;
		[CCode (has_construct_function = false)]
		protected WebRTCICE ();
		public virtual void add_candidate (Gst.WebRTCICEStream stream, string candidate);
		public virtual Gst.WebRTCICEStream? add_stream (uint session_id);
		public virtual bool add_turn_server (string uri);
		public virtual Gst.WebRTCICETransport? find_transport (Gst.WebRTCICEStream stream, Gst.WebRTCICEComponent component);
		public virtual bool gather_candidates (Gst.WebRTCICEStream stream);
		public virtual string get_http_proxy ();
		public virtual bool get_is_controller ();
		[NoWrapper]
		public virtual Gst.WebRTCICECandidateStats get_local_candidates (Gst.WebRTCICEStream stream);
		[NoWrapper]
		public virtual Gst.WebRTCICECandidateStats get_remote_candidates (Gst.WebRTCICEStream stream);
		public virtual bool get_selected_pair (Gst.WebRTCICEStream stream, out Gst.WebRTCICECandidateStats local_stats, out Gst.WebRTCICECandidateStats remote_stats);
		public virtual string? get_stun_server ();
		public virtual string? get_turn_server ();
		public virtual void set_force_relay (bool force_relay);
		public virtual void set_http_proxy (string uri);
		public virtual void set_is_controller (bool controller);
		public virtual bool set_local_credentials (Gst.WebRTCICEStream stream, string ufrag, string pwd);
		public virtual void set_on_ice_candidate (owned Gst.WebRTCICEOnCandidateFunc func);
		public virtual bool set_remote_credentials (Gst.WebRTCICEStream stream, string ufrag, string pwd);
		public virtual void set_stun_server (string? uri);
		public virtual void set_tos (Gst.WebRTCICEStream stream, uint tos);
		public virtual void set_turn_server (string? uri);
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public uint max_rtp_port { get; set construct; }
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public uint min_rtp_port { get; set construct; }
		public signal bool add_local_ip_address (string address);
	}
	[CCode (cheader_filename = "gst/webrtc/webrtc.h", copy_function = "g_boxed_copy", free_function = "g_boxed_free", lower_case_csuffix = "webrtc_ice_candidate_stats", type_id = "gst_webrtc_ice_candidate_stats_get_type ()")]
	[Compact]
	[Version (since = "1.22")]
	public class WebRTCICECandidateStats {
		[CCode (array_length = false)]
		public weak void* _gst_reserved[20];
		public weak string ipaddr;
		public uint port;
		public uint prio;
		public weak string proto;
		public weak string relay_proto;
		public uint stream_id;
		public weak string type;
		public weak string url;
		public Gst.WebRTCICECandidateStats copy ();
		public void free ();
	}
	[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_ice_stream", type_id = "gst_webrtc_ice_stream_get_type ()")]
	[Version (since = "1.22")]
	public abstract class WebRTCICEStream : Gst.Object {
		[CCode (has_construct_function = false)]
		protected WebRTCICEStream ();
		public virtual Gst.WebRTCICETransport? find_transport (Gst.WebRTCICEComponent component);
		public virtual bool gather_candidates ();
		[NoAccessorMethod]
		public uint stream_id { get; construct; }
	}
	[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 sealed class WebRTCRTPReceiver : Gst.Object {
		[CCode (has_construct_function = false)]
		protected WebRTCRTPReceiver ();
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.WebRTCDTLSTransport transport { owned get; }
	}
	[CCode (cheader_filename = "gst/webrtc/webrtc.h", lower_case_csuffix = "webrtc_rtp_sender", type_id = "gst_webrtc_rtp_sender_get_type ()")]
	public sealed class WebRTCRTPSender : Gst.Object {
		[CCode (has_construct_function = false)]
		protected WebRTCRTPSender ();
		[Version (since = "1.20")]
		public void set_priority (Gst.WebRTCPriorityType priority);
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.WebRTCPriorityType priority { get; set; }
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.WebRTCDTLSTransport transport { owned get; }
	}
	[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 (has_construct_function = false)]
		protected WebRTCRTPTransceiver ();
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.Caps codec_preferences { owned get; set; }
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.WebRTCRTPTransceiverDirection current_direction { get; }
		[NoAccessorMethod]
		[Version (since = "1.18")]
		public Gst.WebRTCRTPTransceiverDirection direction { get; set; }
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public Gst.WebRTCKind kind { get; }
		[NoAccessorMethod]
		[Version (since = "1.20")]
		public string mid { owned get; }
		[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", lower_case_csuffix = "webrtc_sctp_transport", type_id = "gst_webrtc_sctp_transport_get_type ()")]
	public abstract class WebRTCSCTPTransport : Gst.Object {
		[CCode (has_construct_function = false)]
		protected WebRTCSCTPTransport ();
		[NoAccessorMethod]
		public uint max_channels { get; }
		[NoAccessorMethod]
		public uint64 max_message_size { get; }
		[NoAccessorMethod]
		public Gst.WebRTCSCTPTransportState state { get; }
		[NoAccessorMethod]
		public Gst.WebRTCDTLSTransport transport { owned get; }
	}
	[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 {
		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 (cname = "gst_webrtc_sdp_type_to_string")]
		public unowned string to_string ();
	}
	[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", cprefix = "GST_WEBRTC_ERROR_", type_id = "gst_webrtc_error_get_type ()")]
	[Version (since = "1.20")]
	public errordomain WebRTCError {
		DATA_CHANNEL_FAILURE,
		DTLS_FAILURE,
		FINGERPRINT_FAILURE,
		SCTP_FAILURE,
		SDP_SYNTAX_ERROR,
		HARDWARE_ENCODER_NOT_AVAILABLE,
		ENCODER_ERROR,
		INVALID_STATE,
		INTERNAL_FAILURE,
		[Version (since = "1.22")]
		INVALID_MODIFICATION,
		[Version (since = "1.22")]
		TYPE_ERROR;
		[CCode (cname = "gst_webrtc_error_quark")]
		public static GLib.Quark quark ();
	}
	[CCode (cheader_filename = "gst/webrtc/webrtc.h", instance_pos = 3.9)]
	[Version (since = "1.22")]
	public delegate void WebRTCICEOnCandidateFunc (Gst.WebRTCICE ice, uint stream_id, string candidate);
	[CCode (cheader_filename = "gst/webrtc/webrtc.h")]
	[Version (replacement = "WebRTCError.quark", since = "1.20")]
	public static GLib.Quark webrtc_error_quark ();
	[CCode (cheader_filename = "gst/webrtc/webrtc.h")]
	[Version (replacement = "WebRTCSDPType.to_string")]
	public static unowned string webrtc_sdp_type_to_string (Gst.WebRTCSDPType type);
}