--- /dev/null
+/* gstreamer-rtsp-server-1.0.vapi generated by lt-vapigen, do not modify. */
+
+[CCode (cprefix = "Gst", gir_namespace = "GstRtspServer", gir_version = "1.0", lower_case_cprefix = "gst_")]
+namespace Gst {
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_auth_get_type ()")]
+ public class RTSPAuth : GLib.Object {
+ [CCode (has_construct_function = false)]
+ public RTSPAuth ();
+ public bool check (Gst.RTSPClient client, GLib.Quark hint, Gst.RTSPClientState state);
+ [NoWrapper]
+ public virtual bool check_method (Gst.RTSPClient client, GLib.Quark hint, Gst.RTSPClientState state);
+ public static string make_basic (string user, string pass);
+ public void set_basic (string basic);
+ public virtual bool setup_auth (Gst.RTSPClient client, GLib.Quark hint, Gst.RTSPClientState state);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_client_get_type ()")]
+ public class RTSPClient : GLib.Object {
+ public weak Gst.RTSPAuth auth;
+ public weak Gst.RTSP.Connection connection;
+ public bool is_ipv6;
+ public weak Gst.RTSPMedia media;
+ public weak Gst.RTSPServer server;
+ public weak string server_ip;
+ public weak GLib.List<void*> sessions;
+ public weak GLib.List<void*> transports;
+ public weak Gst.RTSP.Url uri;
+ public weak Gst.RTSP.Watch watch;
+ public uint watchid;
+ [CCode (has_construct_function = false)]
+ public RTSPClient ();
+ public bool accept (GLib.Socket socket, GLib.Cancellable? cancellable = null) throws GLib.Error;
+ public bool create_from_socket (GLib.Socket socket, string ip, int port, string initial_buffer) throws GLib.Error;
+ public Gst.RTSPAuth get_auth ();
+ public Gst.RTSPMediaMapping get_media_mapping ();
+ public Gst.RTSPServer get_server ();
+ public Gst.RTSPSessionPool get_session_pool ();
+ public bool get_use_client_settings ();
+ public void set_auth (Gst.RTSPAuth auth);
+ public void set_media_mapping (Gst.RTSPMediaMapping mapping);
+ public void set_server (Gst.RTSPServer server);
+ public void set_session_pool (Gst.RTSPSessionPool pool);
+ public void set_use_client_settings (bool use_client_settings);
+ public Gst.RTSPMediaMapping media_mapping { owned get; set; }
+ public Gst.RTSPSessionPool session_pool { owned get; set; }
+ public bool use_client_settings { get; set; }
+ public virtual signal void closed ();
+ public virtual signal void describe_request (void* state);
+ public virtual signal void get_parameter_request (void* state);
+ public virtual signal void new_session (Gst.RTSPSession session);
+ public virtual signal void options_request (void* state);
+ public virtual signal void pause_request (void* state);
+ public virtual signal void play_request (void* state);
+ public virtual signal void set_parameter_request (void* state);
+ public virtual signal void setup_request (void* state);
+ public virtual signal void teardown_request (void* state);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_media_get_type ()")]
+ public class RTSPMedia : GLib.Object {
+ public bool adding;
+ public weak Gst.RTSPAuth auth;
+ public bool buffering;
+ public weak GLib.Cond cond;
+ public weak GLib.List<void*> @dynamic;
+ public weak Gst.Element element;
+ public weak Gst.Element fakesink;
+ public uint id;
+ public bool is_ipv6;
+ public bool is_live;
+ public weak GLib.Mutex @lock;
+ public int n_active;
+ public Gst.Element pipeline;
+ public Gst.RTSP.TimeRange range;
+ public bool reused;
+ public weak Gst.Element rtpbin;
+ public bool seekable;
+ public weak GLib.Source source;
+ public Gst.RTSPMediaStatus status;
+ public weak GLib.GenericArray<void*> streams;
+ public Gst.State target_state;
+ [CCode (has_construct_function = false)]
+ public RTSPMedia ();
+ public void collect_streams ();
+ public unowned Gst.RTSPStream create_stream (Gst.Element payloader, Gst.Pad srcpad);
+ [CCode (vfunc_name = "new_state")]
+ [NoWrapper]
+ public virtual bool emit_new_state (Gst.State state);
+ public Gst.RTSPAuth get_auth ();
+ public uint get_buffer_size ();
+ public uint get_mtu ();
+ public string get_multicast_group ();
+ public Gst.RTSP.LowerTrans get_protocols ();
+ public string get_range_string (bool play);
+ public unowned Gst.RTSPStream get_stream (uint idx);
+ [NoWrapper]
+ public virtual bool handle_message (Gst.Message message);
+ public bool is_eos_shutdown ();
+ public bool is_reusable ();
+ public bool is_shared ();
+ public uint n_streams ();
+ public bool prepare ();
+ public bool seek (Gst.RTSP.TimeRange range);
+ public void set_auth (Gst.RTSPAuth auth);
+ public void set_buffer_size (uint size);
+ public void set_eos_shutdown (bool eos_shutdown);
+ public void set_mtu (uint mtu);
+ public void set_multicast_group (string mc);
+ public void set_protocols (Gst.RTSP.LowerTrans protocols);
+ public void set_reusable (bool reusable);
+ public void set_shared (bool shared);
+ public bool set_state (Gst.State state, GLib.GenericArray<void*> transports);
+ public virtual bool unprepare ();
+ public uint buffer_size { get; set; }
+ [NoAccessorMethod]
+ public bool eos_shutdown { get; set; }
+ public uint mtu { get; set; }
+ public string multicast_group { owned get; set; }
+ public Gst.RTSP.LowerTrans protocols { get; set; }
+ [NoAccessorMethod]
+ public bool reusable { get; set; }
+ [NoAccessorMethod]
+ public bool shared { get; set; }
+ public virtual signal void new_state ();
+ public virtual signal void prepared ();
+ public virtual signal void unprepared ();
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_media_factory_get_type ()")]
+ public class RTSPMediaFactory : GLib.Object {
+ public weak Gst.RTSPAuth auth;
+ public weak GLib.Mutex @lock;
+ public weak GLib.HashTable<void*,void*> medias;
+ public weak GLib.Mutex medias_lock;
+ [CCode (has_construct_function = false)]
+ public RTSPMediaFactory ();
+ [NoWrapper]
+ public virtual void configure (Gst.RTSPMedia media);
+ public virtual Gst.RTSPMedia? @construct (Gst.RTSP.Url url);
+ public virtual unowned Gst.Element create_element (Gst.RTSP.Url url);
+ [NoWrapper]
+ public virtual string gen_key (Gst.RTSP.Url url);
+ public Gst.RTSPAuth get_auth ();
+ public uint get_buffer_size ();
+ public string get_launch ();
+ public string get_multicast_group ();
+ public Gst.RTSP.LowerTrans get_protocols ();
+ public bool is_eos_shutdown ();
+ public bool is_shared ();
+ public void set_auth (Gst.RTSPAuth auth);
+ public void set_buffer_size (uint size);
+ public void set_eos_shutdown (bool eos_shutdown);
+ public void set_launch (string launch);
+ public void set_multicast_group (string mc);
+ public void set_protocols (Gst.RTSP.LowerTrans protocols);
+ public void set_shared (bool shared);
+ public uint buffer_size { get; set; }
+ [NoAccessorMethod]
+ public bool eos_shutdown { get; set; }
+ public string launch { owned get; set; }
+ public string multicast_group { owned get; set; }
+ public Gst.RTSP.LowerTrans protocols { get; set; }
+ [NoAccessorMethod]
+ public bool shared { get; set; }
+ public virtual signal void media_configure (Gst.RTSPMedia media);
+ public virtual signal void media_constructed (Gst.RTSPMedia media);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_media_factory_uri_get_type ()")]
+ public class RTSPMediaFactoryURI : Gst.RTSPMediaFactory {
+ public weak GLib.List<void*> decoders;
+ public weak GLib.List<void*> demuxers;
+ public weak GLib.List<void*> payloaders;
+ public weak Gst.Caps raw_acaps;
+ public weak Gst.Caps raw_vcaps;
+ [CCode (has_construct_function = false)]
+ public RTSPMediaFactoryURI ();
+ public string get_uri ();
+ public void set_uri (string uri);
+ public string uri { owned get; set; }
+ [NoAccessorMethod]
+ public bool use_gstpay { get; set; }
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_media_mapping_get_type ()")]
+ public class RTSPMediaMapping : GLib.Object {
+ public weak GLib.HashTable<void*,void*> mappings;
+ [CCode (has_construct_function = false)]
+ public RTSPMediaMapping ();
+ public void add_factory (string path, owned Gst.RTSPMediaFactory factory);
+ public virtual Gst.RTSPMediaFactory? find_factory (Gst.RTSP.Url url);
+ public void remove_factory (string path);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_server_get_type ()")]
+ public class RTSPServer : GLib.Object {
+ public weak Gst.RTSPAuth auth;
+ public weak GLib.List<void*> clients;
+ public weak GLib.Mutex @lock;
+ public weak GLib.Socket socket;
+ [CCode (has_construct_function = false)]
+ public RTSPServer ();
+ [NoWrapper]
+ public virtual bool accept_client (Gst.RTSPClient client, GLib.Socket socket) throws GLib.Error;
+ public uint attach (GLib.MainContext? context);
+ public GLib.Socket create_socket (GLib.Cancellable? cancellable = null) throws GLib.Error;
+ public GLib.Source create_source (GLib.Cancellable? cancellable = null) throws GLib.Error;
+ public string get_address ();
+ public Gst.RTSPAuth get_auth ();
+ public int get_backlog ();
+ public int get_bound_port ();
+ public Gst.RTSPMediaMapping get_media_mapping ();
+ public string get_service ();
+ public Gst.RTSPSessionPool get_session_pool ();
+ public static bool io_func (GLib.Socket socket, GLib.IOCondition condition, Gst.RTSPServer server);
+ public void set_address (string address);
+ public void set_auth (Gst.RTSPAuth auth);
+ public void set_backlog (int backlog);
+ public void set_media_mapping (Gst.RTSPMediaMapping mapping);
+ public void set_service (string service);
+ public void set_session_pool (Gst.RTSPSessionPool pool);
+ public bool transfer_connection (GLib.Socket socket, string ip, int port, string initial_buffer);
+ public string address { owned get; set; }
+ public int backlog { get; set; }
+ public int bound_port { get; }
+ public Gst.RTSPMediaMapping media_mapping { owned get; set; }
+ public string service { owned get; set; }
+ public Gst.RTSPSessionPool session_pool { owned get; set; }
+ public virtual signal void client_connected (Gst.RTSPClient client);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_session_get_type ()")]
+ public class RTSPSession : GLib.Object {
+ public GLib.TimeVal create_time;
+ public int expire_count;
+ public GLib.TimeVal last_access;
+ public weak GLib.List<void*> medias;
+ [CCode (has_construct_function = false)]
+ public RTSPSession (string sessionid);
+ public void allow_expire ();
+ public unowned Gst.RTSPSessionMedia get_media (Gst.RTSP.Url url);
+ public unowned string get_sessionid ();
+ public uint get_timeout ();
+ public bool is_expired (GLib.TimeVal now);
+ public unowned Gst.RTSPSessionMedia manage_media (Gst.RTSP.Url uri, owned Gst.RTSPMedia media);
+ public int next_timeout (GLib.TimeVal now);
+ public void prevent_expire ();
+ public bool release_media (Gst.RTSPSessionMedia media);
+ public void set_timeout (uint timeout);
+ public void touch ();
+ public string sessionid { get; construct; }
+ public uint timeout { get; set; }
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_session_media_get_type ()")]
+ public class RTSPSessionMedia : GLib.Object {
+ public uint counter;
+ public weak Gst.RTSPMedia media;
+ public Gst.RTSP.State state;
+ public weak GLib.GenericArray<void*> transports;
+ public weak Gst.RTSP.Url url;
+ [CCode (has_construct_function = false)]
+ public RTSPSessionMedia (Gst.RTSP.Url url, Gst.RTSPMedia media);
+ public bool alloc_channels (Gst.RTSP.Range range);
+ public unowned Gst.RTSPStreamTransport get_transport (uint idx);
+ public bool set_state (Gst.State state);
+ public unowned Gst.RTSPStreamTransport set_transport (Gst.RTSPStream stream, Gst.RTSP.Transport tr);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_session_pool_get_type ()")]
+ public class RTSPSessionPool : GLib.Object {
+ public weak GLib.Mutex @lock;
+ public weak GLib.HashTable<void*,void*> sessions;
+ [CCode (has_construct_function = false)]
+ public RTSPSessionPool ();
+ public uint cleanup ();
+ public unowned Gst.RTSPSession create ();
+ [NoWrapper]
+ public virtual string create_session_id ();
+ public GLib.Source create_watch ();
+ public GLib.List<Gst.RTSPSession> filter (Gst.RTSPSessionFilterFunc func);
+ public Gst.RTSPSession find (string sessionid);
+ public uint get_max_sessions ();
+ public uint get_n_sessions ();
+ public bool remove (Gst.RTSPSession sess);
+ public void set_max_sessions (uint max);
+ public uint max_sessions { get; set; }
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_stream_get_type ()")]
+ public class RTSPStream : GLib.Object {
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] appqueue;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] appsink;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] appsrc;
+ public uint buffer_size;
+ public weak Gst.Caps caps;
+ public ulong caps_sig;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] funnel;
+ public uint idx;
+ public bool is_ipv6;
+ public bool is_joined;
+ public uint n_active;
+ public weak Gst.Element payloader;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Pad[] recv_sink;
+ public weak Gst.Pad send_rtp_sink;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Pad[] send_src;
+ public Gst.RTSP.Range server_port;
+ public weak GLib.Object session;
+ public weak Gst.Pad srcpad;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] tee;
+ public weak GLib.List<void*> transports;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] udpsink;
+ [CCode (array_length = false, array_null_terminated = true)]
+ public weak Gst.Element[] udpsrc;
+ [CCode (has_construct_function = false)]
+ public RTSPStream (uint idx, Gst.Element payloader, Gst.Pad srcpad);
+ public bool add_transport (Gst.RTSPStreamTransport trans);
+ public uint get_mtu ();
+ public bool get_rtpinfo (uint rtptime, uint seq);
+ public bool join_bin (Gst.Bin bin, Gst.Element rtpbin, Gst.State state);
+ public bool leave_bin (Gst.Bin bin, Gst.Element rtpbin);
+ public Gst.FlowReturn recv_rtcp (owned Gst.Buffer buffer);
+ public Gst.FlowReturn recv_rtp (owned Gst.Buffer buffer);
+ public bool remove_transport (Gst.RTSPStreamTransport trans);
+ public void set_mtu (uint mtu);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", type_id = "gst_rtsp_stream_transport_get_type ()")]
+ public class RTSPStreamTransport : GLib.Object {
+ public bool active;
+ public weak GLib.DestroyNotify ka_notify;
+ public void* ka_user_data;
+ public weak Gst.RTSPKeepAliveFunc keep_alive;
+ public weak GLib.DestroyNotify notify;
+ public weak GLib.Object rtpsource;
+ public weak Gst.RTSPSendFunc send_rtcp;
+ public weak Gst.RTSPSendFunc send_rtp;
+ public weak Gst.RTSPStream stream;
+ public bool timeout;
+ public Gst.RTSP.Transport transport;
+ public void* user_data;
+ [CCode (has_construct_function = false)]
+ public RTSPStreamTransport (Gst.RTSPStream stream, owned Gst.RTSP.Transport tr);
+ public void set_callbacks (Gst.RTSPSendFunc send_rtp, owned Gst.RTSPSendFunc send_rtcp);
+ public void set_keepalive (owned Gst.RTSPKeepAliveFunc keep_alive);
+ public void set_transport (owned Gst.RTSP.Transport tr);
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", has_type_id = false)]
+ public struct RTSPClientState {
+ public Gst.RTSP.Message request;
+ public weak Gst.RTSP.Url uri;
+ public Gst.RTSP.Method method;
+ public weak Gst.RTSPSession session;
+ public weak Gst.RTSPSessionMedia sessmedia;
+ public weak Gst.RTSPMediaFactory factory;
+ public weak Gst.RTSPMedia media;
+ public weak Gst.RTSPStream stream;
+ public Gst.RTSP.Message response;
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", has_type_id = false)]
+ public struct SDPInfo {
+ public weak string server_proto;
+ public weak string server_ip;
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", cprefix = "GST_RTSP_FILTER_", has_type_id = false)]
+ public enum RTSPFilterResult {
+ REMOVE,
+ KEEP,
+ REF
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", cprefix = "GST_RTSP_MEDIA_STATUS_", has_type_id = false)]
+ public enum RTSPMediaStatus {
+ UNPREPARED,
+ UNPREPARING,
+ PREPARING,
+ PREPARED,
+ ERROR
+ }
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", instance_pos = 0.9)]
+ public delegate void RTSPKeepAliveFunc ();
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", instance_pos = 2.9)]
+ public delegate bool RTSPSendFunc (Gst.Buffer buffer, uint8 channel);
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", instance_pos = 2.9)]
+ public delegate Gst.RTSPFilterResult RTSPSessionFilterFunc (Gst.RTSPSessionPool pool, Gst.RTSPSession session);
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h", instance_pos = 1.9)]
+ public delegate bool RTSPSessionPoolFunc (Gst.RTSPSessionPool pool);
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h")]
+ public static Gst.RTSP.Result rtsp_params_get (Gst.RTSPClient client, Gst.RTSPClientState state);
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h")]
+ public static Gst.RTSP.Result rtsp_params_set (Gst.RTSPClient client, Gst.RTSPClientState state);
+ [CCode (cheader_filename = "gst/rtsp-server/rtsp-server.h")]
+ public static bool rtsp_sdp_from_media (Gst.SDP.Message sdp, Gst.SDPInfo info, Gst.RTSPMedia media);
+}
projects / thirdparty / vala.git / commitdiff
