GstRTSPConnection

This object manages the RTSP connection to the server. It provides function to receive and send bytes and messages.

GstRTSPConnection

Opaque RTSP connection object.


GstRtsp.RTSPConnection

Opaque RTSP connection object.


GstRtsp.RTSPConnection

Opaque RTSP connection object.


Methods

gst_rtsp_connection_clear_auth_params

gst_rtsp_connection_clear_auth_params (GstRTSPConnection * conn)

Clear the list of authentication directives stored in conn.

Parameters:

conn

a GstRTSPConnection


GstRtsp.RTSPConnection.prototype.clear_auth_params

function GstRtsp.RTSPConnection.prototype.clear_auth_params(): {
    // javascript wrapper for 'gst_rtsp_connection_clear_auth_params'
}

Clear the list of authentication directives stored in conn.


GstRtsp.RTSPConnection.clear_auth_params

def GstRtsp.RTSPConnection.clear_auth_params (self):
    #python wrapper for 'gst_rtsp_connection_clear_auth_params'

Clear the list of authentication directives stored in conn.


gst_rtsp_connection_close

GstRTSPResult
gst_rtsp_connection_close (GstRTSPConnection * conn)

Close the connected conn. After this call, the connection is in the same state as when it was first created.

Parameters:

conn

a GstRTSPConnection

Returns

GST_RTSP_OK on success.


GstRtsp.RTSPConnection.prototype.close

function GstRtsp.RTSPConnection.prototype.close(): {
    // javascript wrapper for 'gst_rtsp_connection_close'
}

Close the connected conn. After this call, the connection is in the same state as when it was first created.

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


GstRtsp.RTSPConnection.close

def GstRtsp.RTSPConnection.close (self):
    #python wrapper for 'gst_rtsp_connection_close'

Close the connected conn. After this call, the connection is in the same state as when it was first created.

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


gst_rtsp_connection_connect

GstRTSPResult
gst_rtsp_connection_connect (GstRTSPConnection * conn,
                             GTimeVal * timeout)

Attempt to connect to the url of conn made with gst_rtsp_connection_create. If timeout is NULL this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

timeout

a GTimeVal timeout

Returns

GST_RTSP_OK when a connection could be made.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.connect

function GstRtsp.RTSPConnection.prototype.connect(timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_connect'
}

Attempt to connect to the url of conn made with GstRtsp.prototype.rtsp_connection_create. If timeout is null this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

deprecated : 1.18


GstRtsp.RTSPConnection.connect

def GstRtsp.RTSPConnection.connect (self, timeout):
    #python wrapper for 'gst_rtsp_connection_connect'

Attempt to connect to the url of conn made with GstRtsp.rtsp_connection_create. If timeout is None this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

deprecated : 1.18


gst_rtsp_connection_connect_usec

GstRTSPResult
gst_rtsp_connection_connect_usec (GstRTSPConnection * conn,
                                  gint64 timeout)

Attempt to connect to the url of conn made with gst_rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

timeout

a timeout in microseconds

Returns

GST_RTSP_OK when a connection could be made.

Since : 1.18


GstRtsp.RTSPConnection.prototype.connect_usec

function GstRtsp.RTSPConnection.prototype.connect_usec(timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_connect_usec'
}

Attempt to connect to the url of conn made with GstRtsp.prototype.rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

timeout (Number)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.18


GstRtsp.RTSPConnection.connect_usec

def GstRtsp.RTSPConnection.connect_usec (self, timeout):
    #python wrapper for 'gst_rtsp_connection_connect_usec'

Attempt to connect to the url of conn made with GstRtsp.rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

timeout (int)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.18


gst_rtsp_connection_connect_with_response

GstRTSPResult
gst_rtsp_connection_connect_with_response (GstRTSPConnection * conn,
                                           GTimeVal * timeout,
                                           GstRTSPMessage * response)

Attempt to connect to the url of conn made with gst_rtsp_connection_create. If timeout is NULL this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

timeout

a GTimeVal timeout

response

a GstRTSPMessage

Returns

GST_RTSP_OK when a connection could be made.

Since : 1.8

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.connect_with_response

function GstRtsp.RTSPConnection.prototype.connect_with_response(timeout: GLib.TimeVal, response: GstRtsp.RTSPMessage): {
    // javascript wrapper for 'gst_rtsp_connection_connect_with_response'
}

Attempt to connect to the url of conn made with GstRtsp.prototype.rtsp_connection_create. If timeout is null this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.8

deprecated : 1.18


GstRtsp.RTSPConnection.connect_with_response

def GstRtsp.RTSPConnection.connect_with_response (self, timeout, response):
    #python wrapper for 'gst_rtsp_connection_connect_with_response'

Attempt to connect to the url of conn made with GstRtsp.rtsp_connection_create. If timeout is None this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.8

deprecated : 1.18


gst_rtsp_connection_connect_with_response_usec

GstRTSPResult
gst_rtsp_connection_connect_with_response_usec (GstRTSPConnection * conn,
                                                gint64 timeout,
                                                GstRTSPMessage * response)

Attempt to connect to the url of conn made with gst_rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

timeout

a timeout in microseconds

response

a GstRTSPMessage

Returns

GST_RTSP_OK when a connection could be made.

Since : 1.18


GstRtsp.RTSPConnection.prototype.connect_with_response_usec

function GstRtsp.RTSPConnection.prototype.connect_with_response_usec(timeout: Number, response: GstRtsp.RTSPMessage): {
    // javascript wrapper for 'gst_rtsp_connection_connect_with_response_usec'
}

Attempt to connect to the url of conn made with GstRtsp.prototype.rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

timeout (Number)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.18


GstRtsp.RTSPConnection.connect_with_response_usec

def GstRtsp.RTSPConnection.connect_with_response_usec (self, timeout, response):
    #python wrapper for 'gst_rtsp_connection_connect_with_response_usec'

Attempt to connect to the url of conn made with GstRtsp.rtsp_connection_create. If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired. If conn is set to tunneled, response will contain a response to the tunneling request messages.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

timeout (int)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when a connection could be made.

Since : 1.18


gst_rtsp_connection_do_tunnel

GstRTSPResult
gst_rtsp_connection_do_tunnel (GstRTSPConnection * conn,
                               GstRTSPConnection * conn2)

If conn received the first tunnel connection and conn2 received the second tunnel connection, link the two connections together so that conn manages the tunneled connection.

After this call, conn2 cannot be used anymore and must be freed with gst_rtsp_connection_free.

If conn2 is NULL then only the base64 decoding context will be setup for conn.

Parameters:

conn

a GstRTSPConnection

conn2

a GstRTSPConnection or NULL

Returns

return GST_RTSP_OK on success.


GstRtsp.RTSPConnection.prototype.do_tunnel

function GstRtsp.RTSPConnection.prototype.do_tunnel(conn2: GstRtsp.RTSPConnection): {
    // javascript wrapper for 'gst_rtsp_connection_do_tunnel'
}

If conn received the first tunnel connection and conn2 received the second tunnel connection, link the two connections together so that conn manages the tunneled connection.

After this call, conn2 cannot be used anymore and must be freed with GstRtsp.RTSPConnection.prototype.free.

If conn2 is null then only the base64 decoding context will be setup for conn.

Returns (GstRtsp.RTSPResult)

return GST_RTSP_OK on success.


GstRtsp.RTSPConnection.do_tunnel

def GstRtsp.RTSPConnection.do_tunnel (self, conn2):
    #python wrapper for 'gst_rtsp_connection_do_tunnel'

If conn received the first tunnel connection and conn2 received the second tunnel connection, link the two connections together so that conn manages the tunneled connection.

After this call, conn2 cannot be used anymore and must be freed with GstRtsp.RTSPConnection.free.

If conn2 is None then only the base64 decoding context will be setup for conn.

Returns (GstRtsp.RTSPResult)

return GST_RTSP_OK on success.


gst_rtsp_connection_flush

GstRTSPResult
gst_rtsp_connection_flush (GstRTSPConnection * conn,
                           gboolean flush)

Start or stop the flushing action on conn. When flushing, all current and future actions on conn will return GST_RTSP_EINTR until the connection is set to non-flushing mode again.

Parameters:

conn

a GstRTSPConnection

flush

start or stop the flush

Returns

GST_RTSP_OK.


GstRtsp.RTSPConnection.prototype.flush

function GstRtsp.RTSPConnection.prototype.flush(flush: Number): {
    // javascript wrapper for 'gst_rtsp_connection_flush'
}

Start or stop the flushing action on conn. When flushing, all current and future actions on conn will return GstRtsp.RTSPResult.EINTR until the connection is set to non-flushing mode again.

Parameters:

flush (Number)

start or stop the flush


GstRtsp.RTSPConnection.flush

def GstRtsp.RTSPConnection.flush (self, flush):
    #python wrapper for 'gst_rtsp_connection_flush'

Start or stop the flushing action on conn. When flushing, all current and future actions on conn will return GstRtsp.RTSPResult.EINTR until the connection is set to non-flushing mode again.

Parameters:

flush (bool)

start or stop the flush


gst_rtsp_connection_free

GstRTSPResult
gst_rtsp_connection_free (GstRTSPConnection * conn)

Close and free conn.

Parameters:

conn

a GstRTSPConnection

Returns

GST_RTSP_OK on success.


GstRtsp.RTSPConnection.prototype.free

function GstRtsp.RTSPConnection.prototype.free(): {
    // javascript wrapper for 'gst_rtsp_connection_free'
}

Close and free conn.

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


GstRtsp.RTSPConnection.free

def GstRtsp.RTSPConnection.free (self):
    #python wrapper for 'gst_rtsp_connection_free'

Close and free conn.

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


gst_rtsp_connection_get_ignore_x_server_reply

gboolean
gst_rtsp_connection_get_ignore_x_server_reply (const GstRTSPConnection * conn)

Get the ignore_x_server_reply value.

Parameters:

conn

a GstRTSPConnection

Returns

returns TRUE if the x-server-ip-address header reply will be ignored, else returns FALSE

Since : 1.20


GstRtsp.RTSPConnection.prototype.get_ignore_x_server_reply

function GstRtsp.RTSPConnection.prototype.get_ignore_x_server_reply(): {
    // javascript wrapper for 'gst_rtsp_connection_get_ignore_x_server_reply'
}

Get the ignore_x_server_reply value.

Returns (Number)

returns true if the x-server-ip-address header reply will be ignored, else returns false

Since : 1.20


GstRtsp.RTSPConnection.get_ignore_x_server_reply

def GstRtsp.RTSPConnection.get_ignore_x_server_reply (self):
    #python wrapper for 'gst_rtsp_connection_get_ignore_x_server_reply'

Get the ignore_x_server_reply value.

Returns (bool)

returns True if the x-server-ip-address header reply will be ignored, else returns False

Since : 1.20


gst_rtsp_connection_get_ip

const gchar *
gst_rtsp_connection_get_ip (const GstRTSPConnection * conn)

Retrieve the IP address of the other end of conn.

Parameters:

conn

a GstRTSPConnection

Returns

The IP address as a string. this value remains valid until the connection is closed.


GstRtsp.RTSPConnection.prototype.get_ip

function GstRtsp.RTSPConnection.prototype.get_ip(): {
    // javascript wrapper for 'gst_rtsp_connection_get_ip'
}

Retrieve the IP address of the other end of conn.

Returns (String)

The IP address as a string. this value remains valid until the connection is closed.


GstRtsp.RTSPConnection.get_ip

def GstRtsp.RTSPConnection.get_ip (self):
    #python wrapper for 'gst_rtsp_connection_get_ip'

Retrieve the IP address of the other end of conn.

Returns (str)

The IP address as a string. this value remains valid until the connection is closed.


gst_rtsp_connection_get_read_socket

GSocket *
gst_rtsp_connection_get_read_socket (const GstRTSPConnection * conn)

Get the file descriptor for reading.

Parameters:

conn

a GstRTSPConnection

Returns ( [transfer: none])

the file descriptor used for reading or NULL on error. The file descriptor remains valid until the connection is closed.


GstRtsp.RTSPConnection.prototype.get_read_socket

function GstRtsp.RTSPConnection.prototype.get_read_socket(): {
    // javascript wrapper for 'gst_rtsp_connection_get_read_socket'
}

Get the file descriptor for reading.

Returns (Gio.Socket)

the file descriptor used for reading or null on error. The file descriptor remains valid until the connection is closed.


GstRtsp.RTSPConnection.get_read_socket

def GstRtsp.RTSPConnection.get_read_socket (self):
    #python wrapper for 'gst_rtsp_connection_get_read_socket'

Get the file descriptor for reading.

Returns (Gio.Socket)

the file descriptor used for reading or None on error. The file descriptor remains valid until the connection is closed.


gst_rtsp_connection_get_remember_session_id

gboolean
gst_rtsp_connection_get_remember_session_id (GstRTSPConnection * conn)

Parameters:

conn

a GstRTSPConnection

Returns

TRUE if the GstRTSPConnection remembers the session id in the last response to set it on any further request.


GstRtsp.RTSPConnection.prototype.get_remember_session_id

function GstRtsp.RTSPConnection.prototype.get_remember_session_id(): {
    // javascript wrapper for 'gst_rtsp_connection_get_remember_session_id'
}
Returns (Number)

true if the GstRtsp.RTSPConnection remembers the session id in the last response to set it on any further request.


GstRtsp.RTSPConnection.get_remember_session_id

def GstRtsp.RTSPConnection.get_remember_session_id (self):
    #python wrapper for 'gst_rtsp_connection_get_remember_session_id'
Returns (bool)

True if the GstRtsp.RTSPConnection remembers the session id in the last response to set it on any further request.


gst_rtsp_connection_get_tls

GTlsConnection *
gst_rtsp_connection_get_tls (GstRTSPConnection * conn,
                             GError ** error)

Get the TLS connection of conn.

For client side this will return the GTlsClientConnection when connected over TLS.

For server side connections, this function will create a GTlsServerConnection when called the first time and will return that same connection on subsequent calls. The server is then responsible for configuring the TLS connection.

Parameters:

conn

a GstRTSPConnection

error

GError for error reporting, or NULL to ignore.

Returns ( [transfer: none])

the TLS connection for conn.

Since : 1.2


GstRtsp.RTSPConnection.prototype.get_tls

function GstRtsp.RTSPConnection.prototype.get_tls(): {
    // javascript wrapper for 'gst_rtsp_connection_get_tls'
}

Get the TLS connection of conn.

For client side this will return the Gio.TlsClientConnection when connected over TLS.

For server side connections, this function will create a GTlsServerConnection when called the first time and will return that same connection on subsequent calls. The server is then responsible for configuring the TLS connection.

Returns (Gio.TlsConnection)

the TLS connection for conn.

Since : 1.2


GstRtsp.RTSPConnection.get_tls

@raises(GLib.GError)
def GstRtsp.RTSPConnection.get_tls (self):
    #python wrapper for 'gst_rtsp_connection_get_tls'

Get the TLS connection of conn.

For client side this will return the Gio.TlsClientConnection when connected over TLS.

For server side connections, this function will create a GTlsServerConnection when called the first time and will return that same connection on subsequent calls. The server is then responsible for configuring the TLS connection.

Returns (Gio.TlsConnection)

the TLS connection for conn.

Since : 1.2


gst_rtsp_connection_get_tls_database

GTlsDatabase *
gst_rtsp_connection_get_tls_database (GstRTSPConnection * conn)

Gets the anchor certificate authorities database that will be used after a server certificate can't be verified with the default certificate database.

Parameters:

conn

a GstRTSPConnection

Returns ( [transfer: full])

the anchor certificate authorities database, or NULL if no database has been previously set. Use g_object_unref to release the certificate database.

Since : 1.4


GstRtsp.RTSPConnection.prototype.get_tls_database

function GstRtsp.RTSPConnection.prototype.get_tls_database(): {
    // javascript wrapper for 'gst_rtsp_connection_get_tls_database'
}

Gets the anchor certificate authorities database that will be used after a server certificate can't be verified with the default certificate database.

Returns (Gio.TlsDatabase)

the anchor certificate authorities database, or NULL if no database has been previously set. Use GObject.Object.prototype.unref to release the certificate database.

Since : 1.4


GstRtsp.RTSPConnection.get_tls_database

def GstRtsp.RTSPConnection.get_tls_database (self):
    #python wrapper for 'gst_rtsp_connection_get_tls_database'

Gets the anchor certificate authorities database that will be used after a server certificate can't be verified with the default certificate database.

Returns (Gio.TlsDatabase)

the anchor certificate authorities database, or NULL if no database has been previously set. Use GObject.Object.unref to release the certificate database.

Since : 1.4


gst_rtsp_connection_get_tls_interaction

GTlsInteraction *
gst_rtsp_connection_get_tls_interaction (GstRTSPConnection * conn)

Gets a GTlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Parameters:

conn

a GstRTSPConnection

Returns ( [transfer: full])

a reference on the GTlsInteraction. Use g_object_unref to release.

Since : 1.6


GstRtsp.RTSPConnection.prototype.get_tls_interaction

function GstRtsp.RTSPConnection.prototype.get_tls_interaction(): {
    // javascript wrapper for 'gst_rtsp_connection_get_tls_interaction'
}

Gets a Gio.TlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Returns (Gio.TlsInteraction)

a reference on the Gio.TlsInteraction. Use GObject.Object.prototype.unref to release.

Since : 1.6


GstRtsp.RTSPConnection.get_tls_interaction

def GstRtsp.RTSPConnection.get_tls_interaction (self):
    #python wrapper for 'gst_rtsp_connection_get_tls_interaction'

Gets a Gio.TlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Returns (Gio.TlsInteraction)

a reference on the Gio.TlsInteraction. Use GObject.Object.unref to release.

Since : 1.6


gst_rtsp_connection_get_tls_validation_flags

GTlsCertificateFlags
gst_rtsp_connection_get_tls_validation_flags (GstRTSPConnection * conn)

Gets the TLS validation flags used to verify the peer certificate when a TLS connection is established.

Parameters:

conn

a GstRTSPConnection

Returns

the validationg flags.

Since : 1.2.1


GstRtsp.RTSPConnection.prototype.get_tls_validation_flags

function GstRtsp.RTSPConnection.prototype.get_tls_validation_flags(): {
    // javascript wrapper for 'gst_rtsp_connection_get_tls_validation_flags'
}

Gets the TLS validation flags used to verify the peer certificate when a TLS connection is established.

Returns (Gio.TlsCertificateFlags)

the validationg flags.

Since : 1.2.1


GstRtsp.RTSPConnection.get_tls_validation_flags

def GstRtsp.RTSPConnection.get_tls_validation_flags (self):
    #python wrapper for 'gst_rtsp_connection_get_tls_validation_flags'

Gets the TLS validation flags used to verify the peer certificate when a TLS connection is established.

Returns (Gio.TlsCertificateFlags)

the validationg flags.

Since : 1.2.1


gst_rtsp_connection_get_tunnelid

const gchar *
gst_rtsp_connection_get_tunnelid (const GstRTSPConnection * conn)

Get the tunnel session id the connection.

Parameters:

conn

a GstRTSPConnection

Returns

returns a non-empty string if conn is being tunneled over HTTP.


GstRtsp.RTSPConnection.prototype.get_tunnelid

function GstRtsp.RTSPConnection.prototype.get_tunnelid(): {
    // javascript wrapper for 'gst_rtsp_connection_get_tunnelid'
}

Get the tunnel session id the connection.

Returns (String)

returns a non-empty string if conn is being tunneled over HTTP.


GstRtsp.RTSPConnection.get_tunnelid

def GstRtsp.RTSPConnection.get_tunnelid (self):
    #python wrapper for 'gst_rtsp_connection_get_tunnelid'

Get the tunnel session id the connection.

Returns (str)

returns a non-empty string if conn is being tunneled over HTTP.


gst_rtsp_connection_get_url

GstRTSPUrl *
gst_rtsp_connection_get_url (const GstRTSPConnection * conn)

Retrieve the URL of the other end of conn.

Parameters:

conn

a GstRTSPConnection

Returns

The URL. This value remains valid until the connection is freed.


GstRtsp.RTSPConnection.prototype.get_url

function GstRtsp.RTSPConnection.prototype.get_url(): {
    // javascript wrapper for 'gst_rtsp_connection_get_url'
}

Retrieve the URL of the other end of conn.

Returns (GstRtsp.RTSPUrl)

The URL. This value remains valid until the connection is freed.


GstRtsp.RTSPConnection.get_url

def GstRtsp.RTSPConnection.get_url (self):
    #python wrapper for 'gst_rtsp_connection_get_url'

Retrieve the URL of the other end of conn.

Returns (GstRtsp.RTSPUrl)

The URL. This value remains valid until the connection is freed.


gst_rtsp_connection_get_write_socket

GSocket *
gst_rtsp_connection_get_write_socket (const GstRTSPConnection * conn)

Get the file descriptor for writing.

Parameters:

conn

a GstRTSPConnection

Returns ( [transfer: none])

the file descriptor used for writing or NULL on error. The file descriptor remains valid until the connection is closed.


GstRtsp.RTSPConnection.prototype.get_write_socket

function GstRtsp.RTSPConnection.prototype.get_write_socket(): {
    // javascript wrapper for 'gst_rtsp_connection_get_write_socket'
}

Get the file descriptor for writing.

Returns (Gio.Socket)

the file descriptor used for writing or NULL on error. The file descriptor remains valid until the connection is closed.


GstRtsp.RTSPConnection.get_write_socket

def GstRtsp.RTSPConnection.get_write_socket (self):
    #python wrapper for 'gst_rtsp_connection_get_write_socket'

Get the file descriptor for writing.

Returns (Gio.Socket)

the file descriptor used for writing or NULL on error. The file descriptor remains valid until the connection is closed.


gst_rtsp_connection_is_tunneled

gboolean
gst_rtsp_connection_is_tunneled (const GstRTSPConnection * conn)

Get the tunneling state of the connection.

Parameters:

conn

a GstRTSPConnection

Returns

if conn is using HTTP tunneling.


GstRtsp.RTSPConnection.prototype.is_tunneled

function GstRtsp.RTSPConnection.prototype.is_tunneled(): {
    // javascript wrapper for 'gst_rtsp_connection_is_tunneled'
}

Get the tunneling state of the connection.

Returns (Number)

if conn is using HTTP tunneling.


GstRtsp.RTSPConnection.is_tunneled

def GstRtsp.RTSPConnection.is_tunneled (self):
    #python wrapper for 'gst_rtsp_connection_is_tunneled'

Get the tunneling state of the connection.

Returns (bool)

if conn is using HTTP tunneling.


gst_rtsp_connection_next_timeout

GstRTSPResult
gst_rtsp_connection_next_timeout (GstRTSPConnection * conn,
                                  GTimeVal * timeout)

Calculate the next timeout for conn, storing the result in timeout.

Parameters:

conn

a GstRTSPConnection

timeout

a timeout

Returns

GST_RTSP_OK.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.next_timeout

function GstRtsp.RTSPConnection.prototype.next_timeout(timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_next_timeout'
}

Calculate the next timeout for conn, storing the result in timeout.

Parameters:

timeout (GLib.TimeVal)

a timeout

deprecated : 1.18


GstRtsp.RTSPConnection.next_timeout

def GstRtsp.RTSPConnection.next_timeout (self, timeout):
    #python wrapper for 'gst_rtsp_connection_next_timeout'

Calculate the next timeout for conn, storing the result in timeout.

Parameters:

timeout (GLib.TimeVal)

a timeout

deprecated : 1.18


gst_rtsp_connection_next_timeout_usec

gint64
gst_rtsp_connection_next_timeout_usec (GstRTSPConnection * conn)

Calculate the next timeout for conn

Parameters:

conn

a GstRTSPConnection

Returns

the next timeout in microseconds

Since : 1.18


GstRtsp.RTSPConnection.prototype.next_timeout_usec

function GstRtsp.RTSPConnection.prototype.next_timeout_usec(): {
    // javascript wrapper for 'gst_rtsp_connection_next_timeout_usec'
}

Calculate the next timeout for conn

Returns (Number)

the next timeout in microseconds

Since : 1.18


GstRtsp.RTSPConnection.next_timeout_usec

def GstRtsp.RTSPConnection.next_timeout_usec (self):
    #python wrapper for 'gst_rtsp_connection_next_timeout_usec'

Calculate the next timeout for conn

Returns (int)

the next timeout in microseconds

Since : 1.18


gst_rtsp_connection_poll

GstRTSPResult
gst_rtsp_connection_poll (GstRTSPConnection * conn,
                          GstRTSPEvent events,
                          GstRTSPEvent * revents,
                          GTimeVal * timeout)

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GST_RTSP_OK, revents will contain a bitmask of available operations on conn.

timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

events

a bitmask of GstRTSPEvent flags to check

revents

location for result flags

timeout

a timeout

Returns

GST_RTSP_OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.poll

function GstRtsp.RTSPConnection.prototype.poll(events: GstRtsp.RTSPEvent, revents: GstRtsp.RTSPEvent, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_poll'
}

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GstRtsp.RTSPResult.OK, revents will contain a bitmask of available operations on conn.

timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

events (GstRtsp.RTSPEvent)

a bitmask of GstRtsp.RTSPEvent flags to check

revents (GstRtsp.RTSPEvent)

location for result flags

timeout (GLib.TimeVal)

a timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.poll

def GstRtsp.RTSPConnection.poll (self, events, revents, timeout):
    #python wrapper for 'gst_rtsp_connection_poll'

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GstRtsp.RTSPResult.OK, revents will contain a bitmask of available operations on conn.

timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

events (GstRtsp.RTSPEvent)

a bitmask of GstRtsp.RTSPEvent flags to check

revents (GstRtsp.RTSPEvent)

location for result flags

timeout (GLib.TimeVal)

a timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


gst_rtsp_connection_poll_usec

GstRTSPResult
gst_rtsp_connection_poll_usec (GstRTSPConnection * conn,
                               GstRTSPEvent events,
                               GstRTSPEvent * revents,
                               gint64 timeout)

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GST_RTSP_OK, revents will contain a bitmask of available operations on conn.

timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

events

a bitmask of GstRTSPEvent flags to check

revents

location for result flags

timeout

a timeout in microseconds

Returns

GST_RTSP_OK on success.

Since : 1.18


GstRtsp.RTSPConnection.prototype.poll_usec

function GstRtsp.RTSPConnection.prototype.poll_usec(events: GstRtsp.RTSPEvent, revents: GstRtsp.RTSPEvent, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_poll_usec'
}

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GstRtsp.RTSPResult.OK, revents will contain a bitmask of available operations on conn.

timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

events (GstRtsp.RTSPEvent)

a bitmask of GstRtsp.RTSPEvent flags to check

revents (GstRtsp.RTSPEvent)

location for result flags

timeout (Number)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


GstRtsp.RTSPConnection.poll_usec

def GstRtsp.RTSPConnection.poll_usec (self, events, revents, timeout):
    #python wrapper for 'gst_rtsp_connection_poll_usec'

Wait up to the specified timeout for the connection to become available for at least one of the operations specified in events. When the function returns with GstRtsp.RTSPResult.OK, revents will contain a bitmask of available operations on conn.

timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

events (GstRtsp.RTSPEvent)

a bitmask of GstRtsp.RTSPEvent flags to check

revents (GstRtsp.RTSPEvent)

location for result flags

timeout (int)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


gst_rtsp_connection_read

GstRTSPResult
gst_rtsp_connection_read (GstRTSPConnection * conn,
                          guint8 * data,
                          guint size,
                          GTimeVal * timeout)

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

data

the data to read

size

the size of data

timeout

a timeout value or NULL

Returns

GST_RTSP_OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.read

function GstRtsp.RTSPConnection.prototype.read(data: Number, size: Number, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_read'
}

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

data (Number)

the data to read

size (Number)

the size of data

timeout (GLib.TimeVal)

a timeout value or null

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.read

def GstRtsp.RTSPConnection.read (self, data, size, timeout):
    #python wrapper for 'gst_rtsp_connection_read'

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

data (int)

the data to read

size (int)

the size of data

timeout (GLib.TimeVal)

a timeout value or None

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


gst_rtsp_connection_read_usec

GstRTSPResult
gst_rtsp_connection_read_usec (GstRTSPConnection * conn,
                               guint8 * data,
                               guint size,
                               gint64 timeout)

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

data

the data to read

size

the size of data

timeout

a timeout value in microseconds

Returns

GST_RTSP_OK on success.

Since : 1.18


GstRtsp.RTSPConnection.prototype.read_usec

function GstRtsp.RTSPConnection.prototype.read_usec(data: Number, size: Number, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_read_usec'
}

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

data (Number)

the data to read

size (Number)

the size of data

timeout (Number)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


GstRtsp.RTSPConnection.read_usec

def GstRtsp.RTSPConnection.read_usec (self, data, size, timeout):
    #python wrapper for 'gst_rtsp_connection_read_usec'

Attempt to read size bytes into data from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

data (int)

the data to read

size (int)

the size of data

timeout (int)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


gst_rtsp_connection_receive

GstRTSPResult
gst_rtsp_connection_receive (GstRTSPConnection * conn,
                             GstRTSPMessage * message,
                             GTimeVal * timeout)

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

message

the message to read

timeout

a timeout value or NULL

Returns

GST_RTSP_OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.receive

function GstRtsp.RTSPConnection.prototype.receive(message: GstRtsp.RTSPMessage, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_receive'
}

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to read

timeout (GLib.TimeVal)

a timeout value or null

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.receive

def GstRtsp.RTSPConnection.receive (self, message, timeout):
    #python wrapper for 'gst_rtsp_connection_receive'

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to read

timeout (GLib.TimeVal)

a timeout value or None

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


gst_rtsp_connection_receive_usec

GstRTSPResult
gst_rtsp_connection_receive_usec (GstRTSPConnection * conn,
                                  GstRTSPMessage * message,
                                  gint64 timeout)

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

message

the message to read

timeout

a timeout value or 0

Returns

GST_RTSP_OK on success.

Since : 1.18


GstRtsp.RTSPConnection.prototype.receive_usec

function GstRtsp.RTSPConnection.prototype.receive_usec(message: GstRtsp.RTSPMessage, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_receive_usec'
}

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to read

timeout (Number)

a timeout value or 0

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


GstRtsp.RTSPConnection.receive_usec

def GstRtsp.RTSPConnection.receive_usec (self, message, timeout):
    #python wrapper for 'gst_rtsp_connection_receive_usec'

Attempt to read into message from the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to read

timeout (int)

a timeout value or 0

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


gst_rtsp_connection_reset_timeout

GstRTSPResult
gst_rtsp_connection_reset_timeout (GstRTSPConnection * conn)

Reset the timeout of conn.

Parameters:

conn

a GstRTSPConnection

Returns

GST_RTSP_OK.


GstRtsp.RTSPConnection.prototype.reset_timeout

function GstRtsp.RTSPConnection.prototype.reset_timeout(): {
    // javascript wrapper for 'gst_rtsp_connection_reset_timeout'
}

Reset the timeout of conn.


GstRtsp.RTSPConnection.reset_timeout

def GstRtsp.RTSPConnection.reset_timeout (self):
    #python wrapper for 'gst_rtsp_connection_reset_timeout'

Reset the timeout of conn.


gst_rtsp_connection_send

GstRTSPResult
gst_rtsp_connection_send (GstRTSPConnection * conn,
                          GstRTSPMessage * message,
                          GTimeVal * timeout)

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

message

the message to send

timeout

a timeout value or NULL

Returns

GST_RTSP_OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.send

function GstRtsp.RTSPConnection.prototype.send(message: GstRtsp.RTSPMessage, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_send'
}

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to send

timeout (GLib.TimeVal)

a timeout value or null

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.send

def GstRtsp.RTSPConnection.send (self, message, timeout):
    #python wrapper for 'gst_rtsp_connection_send'

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to send

timeout (GLib.TimeVal)

a timeout value or None

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


gst_rtsp_connection_send_messages

GstRTSPResult
gst_rtsp_connection_send_messages (GstRTSPConnection * conn,
                                   GstRTSPMessage * messages,
                                   guint n_messages,
                                   GTimeVal * timeout)

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

messages ( [arraylength=n_messages])

the messages to send

n_messages

the number of messages to send

timeout

a timeout value or NULL

Returns

GST_RTSP_OK on success.

Since : 1.16

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.send_messages

function GstRtsp.RTSPConnection.prototype.send_messages(messages: [ GstRtsp.RTSPMessage ], n_messages: Number, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_send_messages'
}

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (Number)

the number of messages to send

timeout (GLib.TimeVal)

a timeout value or null

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.16

deprecated : 1.18


GstRtsp.RTSPConnection.send_messages

def GstRtsp.RTSPConnection.send_messages (self, messages, n_messages, timeout):
    #python wrapper for 'gst_rtsp_connection_send_messages'

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (int)

the number of messages to send

timeout (GLib.TimeVal)

a timeout value or None

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.16

deprecated : 1.18


gst_rtsp_connection_send_messages_usec

GstRTSPResult
gst_rtsp_connection_send_messages_usec (GstRTSPConnection * conn,
                                        GstRTSPMessage * messages,
                                        guint n_messages,
                                        gint64 timeout)

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

messages ( [arraylength=n_messages])

the messages to send

n_messages

the number of messages to send

timeout

a timeout value in microseconds

Returns

GST_RTSP_OK on Since.

Since : 1.18


GstRtsp.RTSPConnection.prototype.send_messages_usec

function GstRtsp.RTSPConnection.prototype.send_messages_usec(messages: [ GstRtsp.RTSPMessage ], n_messages: Number, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_send_messages_usec'
}

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (Number)

the number of messages to send

timeout (Number)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on Since.

Since : 1.18


GstRtsp.RTSPConnection.send_messages_usec

def GstRtsp.RTSPConnection.send_messages_usec (self, messages, n_messages, timeout):
    #python wrapper for 'gst_rtsp_connection_send_messages_usec'

Attempt to send messages to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (int)

the number of messages to send

timeout (int)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on Since.

Since : 1.18


gst_rtsp_connection_send_usec

GstRTSPResult
gst_rtsp_connection_send_usec (GstRTSPConnection * conn,
                               GstRTSPMessage * message,
                               gint64 timeout)

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

message

the message to send

timeout

a timeout value in microseconds

Returns

GST_RTSP_OK on success.

Since : 1.18


GstRtsp.RTSPConnection.prototype.send_usec

function GstRtsp.RTSPConnection.prototype.send_usec(message: GstRtsp.RTSPMessage, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_send_usec'
}

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to send

timeout (Number)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


GstRtsp.RTSPConnection.send_usec

def GstRtsp.RTSPConnection.send_usec (self, message, timeout):
    #python wrapper for 'gst_rtsp_connection_send_usec'

Attempt to send message to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

message (GstRtsp.RTSPMessage)

the message to send

timeout (int)

a timeout value in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


gst_rtsp_connection_set_accept_certificate_func

gst_rtsp_connection_set_accept_certificate_func (GstRTSPConnection * conn,
                                                 GstRTSPConnectionAcceptCertificateFunc func,
                                                 gpointer user_data,
                                                 GDestroyNotify destroy_notify)

Sets a custom accept-certificate function for checking certificates for validity. This will directly map to GTlsConnection 's "accept-certificate" signal and be performed after the default checks of GstRTSPConnection (checking against the GTlsDatabase with the given GTlsCertificateFlags) have failed. If no GTlsDatabase is set on this connection, only func will be called.

Parameters:

conn

a GstRTSPConnection

func

a GstRTSPConnectionAcceptCertificateFunc to check certificates

user_data

User data passed to func

destroy_notify

GDestroyNotify for user_data

Since : 1.14


GstRtsp.RTSPConnection.prototype.set_accept_certificate_func

function GstRtsp.RTSPConnection.prototype.set_accept_certificate_func(func: GstRtsp.RTSPConnectionAcceptCertificateFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_connection_set_accept_certificate_func'
}

Sets a custom accept-certificate function for checking certificates for validity. This will directly map to Gio.TlsConnection 's "accept-certificate" signal and be performed after the default checks of GstRtsp.RTSPConnection (checking against the Gio.TlsDatabase with the given Gio.TlsCertificateFlags) have failed. If no Gio.TlsDatabase is set on this connection, only func will be called.

Parameters:

user_data (Object)

User data passed to func

Since : 1.14


GstRtsp.RTSPConnection.set_accept_certificate_func

def GstRtsp.RTSPConnection.set_accept_certificate_func (self, func, *user_data):
    #python wrapper for 'gst_rtsp_connection_set_accept_certificate_func'

Sets a custom accept-certificate function for checking certificates for validity. This will directly map to Gio.TlsConnection 's "accept-certificate" signal and be performed after the default checks of GstRtsp.RTSPConnection (checking against the Gio.TlsDatabase with the given Gio.TlsCertificateFlags) have failed. If no Gio.TlsDatabase is set on this connection, only func will be called.

Parameters:

user_data (variadic)

User data passed to func

Since : 1.14


gst_rtsp_connection_set_auth

GstRTSPResult
gst_rtsp_connection_set_auth (GstRTSPConnection * conn,
                              GstRTSPAuthMethod method,
                              const gchar * user,
                              const gchar * pass)

Configure conn for authentication mode method with user and pass as the user and password respectively.

Parameters:

conn

a GstRTSPConnection

method

authentication method

user

the user

pass

the password

Returns

GST_RTSP_OK.


GstRtsp.RTSPConnection.prototype.set_auth

function GstRtsp.RTSPConnection.prototype.set_auth(method: GstRtsp.RTSPAuthMethod, user: String, pass: String): {
    // javascript wrapper for 'gst_rtsp_connection_set_auth'
}

Configure conn for authentication mode method with user and pass as the user and password respectively.

Parameters:

method (GstRtsp.RTSPAuthMethod)

authentication method

user (String)

the user

pass (String)

the password


GstRtsp.RTSPConnection.set_auth

def GstRtsp.RTSPConnection.set_auth (self, method, user, pass):
    #python wrapper for 'gst_rtsp_connection_set_auth'

Configure conn for authentication mode method with user and pass as the user and password respectively.

Parameters:

method (GstRtsp.RTSPAuthMethod)

authentication method

user (str)

the user

pass (str)

the password


gst_rtsp_connection_set_auth_param

gst_rtsp_connection_set_auth_param (GstRTSPConnection * conn,
                                    const gchar * param,
                                    const gchar * value)

Setup conn with authentication directives. This is not necessary for methods GST_RTSP_AUTH_NONE and GST_RTSP_AUTH_BASIC. For GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge in the WWW-Authenticate response header and can include realm, domain, nonce, opaque, stale, algorithm, qop as per RFC2617.

Parameters:

conn

a GstRTSPConnection

param

authentication directive

value

value


GstRtsp.RTSPConnection.prototype.set_auth_param

function GstRtsp.RTSPConnection.prototype.set_auth_param(param: String, value: String): {
    // javascript wrapper for 'gst_rtsp_connection_set_auth_param'
}

Setup conn with authentication directives. This is not necessary for methods GstRtsp.RTSPAuthMethod.NONE and GstRtsp.RTSPAuthMethod.BASIC. For GstRtsp.RTSPAuthMethod.DIGEST, directives should be taken from the digest challenge in the WWW-Authenticate response header and can include realm, domain, nonce, opaque, stale, algorithm, qop as per RFC2617.

Parameters:

param (String)

authentication directive

value (String)

value


GstRtsp.RTSPConnection.set_auth_param

def GstRtsp.RTSPConnection.set_auth_param (self, param, value):
    #python wrapper for 'gst_rtsp_connection_set_auth_param'

Setup conn with authentication directives. This is not necessary for methods GstRtsp.RTSPAuthMethod.NONE and GstRtsp.RTSPAuthMethod.BASIC. For GstRtsp.RTSPAuthMethod.DIGEST, directives should be taken from the digest challenge in the WWW-Authenticate response header and can include realm, domain, nonce, opaque, stale, algorithm, qop as per RFC2617.

Parameters:

param (str)

authentication directive

value (str)

value


gst_rtsp_connection_set_content_length_limit

gst_rtsp_connection_set_content_length_limit (GstRTSPConnection * conn,
                                              guint limit)

Configure conn to use the specified Content-Length limit. Both requests and responses are validated. If content-length is exceeded, ENOMEM error will be returned.

Parameters:

conn

a GstRTSPConnection

limit

Content-Length limit

Since : 1.18


GstRtsp.RTSPConnection.prototype.set_content_length_limit

function GstRtsp.RTSPConnection.prototype.set_content_length_limit(limit: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_content_length_limit'
}

Configure conn to use the specified Content-Length limit. Both requests and responses are validated. If content-length is exceeded, ENOMEM error will be returned.

Parameters:

limit (Number)

Content-Length limit

Since : 1.18


GstRtsp.RTSPConnection.set_content_length_limit

def GstRtsp.RTSPConnection.set_content_length_limit (self, limit):
    #python wrapper for 'gst_rtsp_connection_set_content_length_limit'

Configure conn to use the specified Content-Length limit. Both requests and responses are validated. If content-length is exceeded, ENOMEM error will be returned.

Parameters:

limit (int)

Content-Length limit

Since : 1.18


gst_rtsp_connection_set_http_mode

gst_rtsp_connection_set_http_mode (GstRTSPConnection * conn,
                                   gboolean enable)

By setting the HTTP mode to TRUE the message parsing will support HTTP messages in addition to the RTSP messages. It will also disable the automatic handling of setting up an HTTP tunnel.

Parameters:

conn

a GstRTSPConnection

enable

TRUE to enable manual HTTP mode


GstRtsp.RTSPConnection.prototype.set_http_mode

function GstRtsp.RTSPConnection.prototype.set_http_mode(enable: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_http_mode'
}

By setting the HTTP mode to true the message parsing will support HTTP messages in addition to the RTSP messages. It will also disable the automatic handling of setting up an HTTP tunnel.

Parameters:

enable (Number)

true to enable manual HTTP mode


GstRtsp.RTSPConnection.set_http_mode

def GstRtsp.RTSPConnection.set_http_mode (self, enable):
    #python wrapper for 'gst_rtsp_connection_set_http_mode'

By setting the HTTP mode to True the message parsing will support HTTP messages in addition to the RTSP messages. It will also disable the automatic handling of setting up an HTTP tunnel.

Parameters:

enable (bool)

True to enable manual HTTP mode


gst_rtsp_connection_set_ignore_x_server_reply

gst_rtsp_connection_set_ignore_x_server_reply (GstRTSPConnection * conn,
                                               gboolean ignore)

Set whether to ignore the x-server-ip-address header reply or not. If the header is ignored, the original address will be used instead.

Parameters:

conn

a GstRTSPConnection

ignore

TRUE to ignore the x-server-ip-address header reply or FALSE to comply with it (%FALSE is the default).

Since : 1.20


GstRtsp.RTSPConnection.prototype.set_ignore_x_server_reply

function GstRtsp.RTSPConnection.prototype.set_ignore_x_server_reply(ignore: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_ignore_x_server_reply'
}

Set whether to ignore the x-server-ip-address header reply or not. If the header is ignored, the original address will be used instead.

Parameters:

ignore (Number)

true to ignore the x-server-ip-address header reply or false to comply with it (%FALSE is the default).

Since : 1.20


GstRtsp.RTSPConnection.set_ignore_x_server_reply

def GstRtsp.RTSPConnection.set_ignore_x_server_reply (self, ignore):
    #python wrapper for 'gst_rtsp_connection_set_ignore_x_server_reply'

Set whether to ignore the x-server-ip-address header reply or not. If the header is ignored, the original address will be used instead.

Parameters:

ignore (bool)

True to ignore the x-server-ip-address header reply or False to comply with it (%FALSE is the default).

Since : 1.20


gst_rtsp_connection_set_ip

gst_rtsp_connection_set_ip (GstRTSPConnection * conn,
                            const gchar * ip)

Set the IP address of the server.

Parameters:

conn

a GstRTSPConnection

ip

an ip address


GstRtsp.RTSPConnection.prototype.set_ip

function GstRtsp.RTSPConnection.prototype.set_ip(ip: String): {
    // javascript wrapper for 'gst_rtsp_connection_set_ip'
}

Set the IP address of the server.

Parameters:

ip (String)

an ip address


GstRtsp.RTSPConnection.set_ip

def GstRtsp.RTSPConnection.set_ip (self, ip):
    #python wrapper for 'gst_rtsp_connection_set_ip'

Set the IP address of the server.

Parameters:

ip (str)

an ip address


gst_rtsp_connection_set_proxy

GstRTSPResult
gst_rtsp_connection_set_proxy (GstRTSPConnection * conn,
                               const gchar * host,
                               guint port)

Set the proxy host and port.

Parameters:

conn

a GstRTSPConnection

host

the proxy host

port

the proxy port

Returns

GST_RTSP_OK.


GstRtsp.RTSPConnection.prototype.set_proxy

function GstRtsp.RTSPConnection.prototype.set_proxy(host: String, port: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_proxy'
}

Set the proxy host and port.

Parameters:

host (String)

the proxy host

port (Number)

the proxy port


GstRtsp.RTSPConnection.set_proxy

def GstRtsp.RTSPConnection.set_proxy (self, host, port):
    #python wrapper for 'gst_rtsp_connection_set_proxy'

Set the proxy host and port.

Parameters:

host (str)

the proxy host

port (int)

the proxy port


gst_rtsp_connection_set_qos_dscp

GstRTSPResult
gst_rtsp_connection_set_qos_dscp (GstRTSPConnection * conn,
                                  guint qos_dscp)

Configure conn to use the specified DSCP value.

Parameters:

conn

a GstRTSPConnection

qos_dscp

DSCP value

Returns

GST_RTSP_OK on success.


GstRtsp.RTSPConnection.prototype.set_qos_dscp

function GstRtsp.RTSPConnection.prototype.set_qos_dscp(qos_dscp: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_qos_dscp'
}

Configure conn to use the specified DSCP value.

Parameters:

qos_dscp (Number)

DSCP value

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


GstRtsp.RTSPConnection.set_qos_dscp

def GstRtsp.RTSPConnection.set_qos_dscp (self, qos_dscp):
    #python wrapper for 'gst_rtsp_connection_set_qos_dscp'

Configure conn to use the specified DSCP value.

Parameters:

qos_dscp (int)

DSCP value

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.


gst_rtsp_connection_set_remember_session_id

gst_rtsp_connection_set_remember_session_id (GstRTSPConnection * conn,
                                             gboolean remember)

Sets if the GstRTSPConnection should remember the session id from the last response received and force it onto any further requests.

The default value is TRUE

Parameters:

conn

a GstRTSPConnection

remember

TRUE if the connection should remember the session id


GstRtsp.RTSPConnection.prototype.set_remember_session_id

function GstRtsp.RTSPConnection.prototype.set_remember_session_id(remember: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_remember_session_id'
}

Sets if the GstRtsp.RTSPConnection should remember the session id from the last response received and force it onto any further requests.

The default value is true

Parameters:

remember (Number)

true if the connection should remember the session id


GstRtsp.RTSPConnection.set_remember_session_id

def GstRtsp.RTSPConnection.set_remember_session_id (self, remember):
    #python wrapper for 'gst_rtsp_connection_set_remember_session_id'

Sets if the GstRtsp.RTSPConnection should remember the session id from the last response received and force it onto any further requests.

The default value is True

Parameters:

remember (bool)

True if the connection should remember the session id


gst_rtsp_connection_set_tls_database

gst_rtsp_connection_set_tls_database (GstRTSPConnection * conn,
                                      GTlsDatabase * database)

Sets the anchor certificate authorities database. This certificate database will be used to verify the server's certificate in case it can't be verified with the default certificate database first.

Parameters:

conn

a GstRTSPConnection

database

a GTlsDatabase

Since : 1.4


GstRtsp.RTSPConnection.prototype.set_tls_database

function GstRtsp.RTSPConnection.prototype.set_tls_database(database: Gio.TlsDatabase): {
    // javascript wrapper for 'gst_rtsp_connection_set_tls_database'
}

Sets the anchor certificate authorities database. This certificate database will be used to verify the server's certificate in case it can't be verified with the default certificate database first.

Since : 1.4


GstRtsp.RTSPConnection.set_tls_database

def GstRtsp.RTSPConnection.set_tls_database (self, database):
    #python wrapper for 'gst_rtsp_connection_set_tls_database'

Sets the anchor certificate authorities database. This certificate database will be used to verify the server's certificate in case it can't be verified with the default certificate database first.

Since : 1.4


gst_rtsp_connection_set_tls_interaction

gst_rtsp_connection_set_tls_interaction (GstRTSPConnection * conn,
                                         GTlsInteraction * interaction)

Sets a GTlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Parameters:

conn

a GstRTSPConnection

interaction

a GTlsInteraction

Since : 1.6


GstRtsp.RTSPConnection.prototype.set_tls_interaction

function GstRtsp.RTSPConnection.prototype.set_tls_interaction(interaction: Gio.TlsInteraction): {
    // javascript wrapper for 'gst_rtsp_connection_set_tls_interaction'
}

Sets a Gio.TlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Since : 1.6


GstRtsp.RTSPConnection.set_tls_interaction

def GstRtsp.RTSPConnection.set_tls_interaction (self, interaction):
    #python wrapper for 'gst_rtsp_connection_set_tls_interaction'

Sets a Gio.TlsInteraction object to be used when the connection or certificate database need to interact with the user. This will be used to prompt the user for passwords where necessary.

Since : 1.6


gst_rtsp_connection_set_tls_validation_flags

gboolean
gst_rtsp_connection_set_tls_validation_flags (GstRTSPConnection * conn,
                                              GTlsCertificateFlags flags)

Sets the TLS validation flags to be used to verify the peer certificate when a TLS connection is established.

Parameters:

conn

a GstRTSPConnection

flags

the validation flags.

Returns

TRUE if the validation flags are set correctly, or FALSE if conn is NULL or is not a TLS connection.

Since : 1.2.1


GstRtsp.RTSPConnection.prototype.set_tls_validation_flags

function GstRtsp.RTSPConnection.prototype.set_tls_validation_flags(flags: Gio.TlsCertificateFlags): {
    // javascript wrapper for 'gst_rtsp_connection_set_tls_validation_flags'
}

Sets the TLS validation flags to be used to verify the peer certificate when a TLS connection is established.

Parameters:

the validation flags.

Returns (Number)

TRUE if the validation flags are set correctly, or FALSE if conn is NULL or is not a TLS connection.

Since : 1.2.1


GstRtsp.RTSPConnection.set_tls_validation_flags

def GstRtsp.RTSPConnection.set_tls_validation_flags (self, flags):
    #python wrapper for 'gst_rtsp_connection_set_tls_validation_flags'

Sets the TLS validation flags to be used to verify the peer certificate when a TLS connection is established.

Parameters:

the validation flags.

Returns (bool)

TRUE if the validation flags are set correctly, or FALSE if conn is NULL or is not a TLS connection.

Since : 1.2.1


gst_rtsp_connection_set_tunneled

gst_rtsp_connection_set_tunneled (GstRTSPConnection * conn,
                                  gboolean tunneled)

Set the HTTP tunneling state of the connection. This must be configured before the conn is connected.

Parameters:

conn

a GstRTSPConnection

tunneled

the new state


GstRtsp.RTSPConnection.prototype.set_tunneled

function GstRtsp.RTSPConnection.prototype.set_tunneled(tunneled: Number): {
    // javascript wrapper for 'gst_rtsp_connection_set_tunneled'
}

Set the HTTP tunneling state of the connection. This must be configured before the conn is connected.

Parameters:

tunneled (Number)

the new state


GstRtsp.RTSPConnection.set_tunneled

def GstRtsp.RTSPConnection.set_tunneled (self, tunneled):
    #python wrapper for 'gst_rtsp_connection_set_tunneled'

Set the HTTP tunneling state of the connection. This must be configured before the conn is connected.

Parameters:

tunneled (bool)

the new state


gst_rtsp_connection_write

GstRTSPResult
gst_rtsp_connection_write (GstRTSPConnection * conn,
                           const guint8 * data,
                           guint size,
                           GTimeVal * timeout)

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be NULL, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

data

the data to write

size

the size of data

timeout

a timeout value or NULL

Returns

GST_RTSP_OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.prototype.write

function GstRtsp.RTSPConnection.prototype.write(data: Number, size: Number, timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_connection_write'
}

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be null, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

data (Number)

the data to write

size (Number)

the size of data

timeout (GLib.TimeVal)

a timeout value or null

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


GstRtsp.RTSPConnection.write

def GstRtsp.RTSPConnection.write (self, data, size, timeout):
    #python wrapper for 'gst_rtsp_connection_write'

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be None, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

data (int)

the data to write

size (int)

the size of data

timeout (GLib.TimeVal)

a timeout value or None

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

deprecated : 1.18


gst_rtsp_connection_write_usec

GstRTSPResult
gst_rtsp_connection_write_usec (GstRTSPConnection * conn,
                                const guint8 * data,
                                guint size,
                                gint64 timeout)

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with gst_rtsp_connection_flush.

Parameters:

conn

a GstRTSPConnection

data

the data to write

size

the size of data

timeout

a timeout value or 0

Returns

GST_RTSP_OK on success.

Since : 1.18


GstRtsp.RTSPConnection.prototype.write_usec

function GstRtsp.RTSPConnection.prototype.write_usec(data: Number, size: Number, timeout: Number): {
    // javascript wrapper for 'gst_rtsp_connection_write_usec'
}

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.prototype.flush.

Parameters:

data (Number)

the data to write

size (Number)

the size of data

timeout (Number)

a timeout value or 0

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


GstRtsp.RTSPConnection.write_usec

def GstRtsp.RTSPConnection.write_usec (self, data, size, timeout):
    #python wrapper for 'gst_rtsp_connection_write_usec'

Attempt to write size bytes of data to the connected conn, blocking up to the specified timeout. timeout can be 0, in which case this function might block forever.

This function can be cancelled with GstRtsp.RTSPConnection.flush.

Parameters:

data (int)

the data to write

size (int)

the size of data

timeout (int)

a timeout value or 0

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK on success.

Since : 1.18


Functions

gst_rtsp_connection_accept

GstRTSPResult
gst_rtsp_connection_accept (GSocket * socket,
                            GstRTSPConnection ** conn,
                            GCancellable * cancellable)

Accept a new connection on socket and create a new GstRTSPConnection for handling communication on new socket.

Parameters:

socket

a socket

conn ( [out][transfer: full])

storage for a GstRTSPConnection

cancellable

a GCancellable to cancel the operation

Returns

GST_RTSP_OK when conn contains a valid connection.


GstRtsp.prototype.rtsp_connection_accept

function GstRtsp.prototype.rtsp_connection_accept(socket: Gio.Socket, cancellable: Gio.Cancellable): {
    // javascript wrapper for 'gst_rtsp_connection_accept'
}

Accept a new connection on socket and create a new GstRtsp.RTSPConnection for handling communication on new socket.

Parameters:

socket (Gio.Socket)

a socket

cancellable (Gio.Cancellable)

a Gio.Cancellable to cancel the operation

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


GstRtsp.rtsp_connection_accept

def GstRtsp.rtsp_connection_accept (socket, cancellable):
    #python wrapper for 'gst_rtsp_connection_accept'

Accept a new connection on socket and create a new GstRtsp.RTSPConnection for handling communication on new socket.

Parameters:

socket (Gio.Socket)

a socket

cancellable (Gio.Cancellable)

a Gio.Cancellable to cancel the operation

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


gst_rtsp_connection_create

GstRTSPResult
gst_rtsp_connection_create (const GstRTSPUrl * url,
                            GstRTSPConnection ** conn)

Create a newly allocated GstRTSPConnection from url and store it in conn. The connection will not yet attempt to connect to url, use gst_rtsp_connection_connect.

A copy of url will be made.

Parameters:

url

a GstRTSPUrl

conn ( [out][transfer: full])

storage for a GstRTSPConnection

Returns

GST_RTSP_OK when conn contains a valid connection.


GstRtsp.prototype.rtsp_connection_create

function GstRtsp.prototype.rtsp_connection_create(url: GstRtsp.RTSPUrl): {
    // javascript wrapper for 'gst_rtsp_connection_create'
}

Create a newly allocated GstRtsp.RTSPConnection from url and store it in conn. The connection will not yet attempt to connect to url, use GstRtsp.RTSPConnection.prototype.connect.

A copy of url will be made.

Parameters:

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


GstRtsp.rtsp_connection_create

def GstRtsp.rtsp_connection_create (url):
    #python wrapper for 'gst_rtsp_connection_create'

Create a newly allocated GstRtsp.RTSPConnection from url and store it in conn. The connection will not yet attempt to connect to url, use GstRtsp.RTSPConnection.connect.

A copy of url will be made.

Parameters:

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


gst_rtsp_connection_create_from_socket

GstRTSPResult
gst_rtsp_connection_create_from_socket (GSocket * socket,
                                        const gchar * ip,
                                        guint16 port,
                                        const gchar * initial_buffer,
                                        GstRTSPConnection ** conn)

Create a new GstRTSPConnection for handling communication on the existing socket socket. The initial_buffer contains zero terminated data already read from socket which should be used before starting to read new data.

Parameters:

socket

a GSocket

ip

the IP address of the other end

port

the port used by the other end

initial_buffer

data already read from fd

conn ( [out][transfer: full])

storage for a GstRTSPConnection

Returns

GST_RTSP_OK when conn contains a valid connection.


GstRtsp.prototype.rtsp_connection_create_from_socket

function GstRtsp.prototype.rtsp_connection_create_from_socket(socket: Gio.Socket, ip: String, port: Number, initial_buffer: String): {
    // javascript wrapper for 'gst_rtsp_connection_create_from_socket'
}

Create a new GstRtsp.RTSPConnection for handling communication on the existing socket socket. The initial_buffer contains zero terminated data already read from socket which should be used before starting to read new data.

Parameters:

socket (Gio.Socket)

a Gio.Socket

ip (String)

the IP address of the other end

port (Number)

the port used by the other end

initial_buffer (String)

data already read from fd

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


GstRtsp.rtsp_connection_create_from_socket

def GstRtsp.rtsp_connection_create_from_socket (socket, ip, port, initial_buffer):
    #python wrapper for 'gst_rtsp_connection_create_from_socket'

Create a new GstRtsp.RTSPConnection for handling communication on the existing socket socket. The initial_buffer contains zero terminated data already read from socket which should be used before starting to read new data.

Parameters:

socket (Gio.Socket)

a Gio.Socket

ip (str)

the IP address of the other end

port (int)

the port used by the other end

initial_buffer (str)

data already read from fd

Returns a tuple made of:

GstRtsp.RTSPResult.OK when conn contains a valid connection.

GstRtsp.RTSPResult.OK when conn contains a valid connection.


GstRTSPWatch

Opaque RTSP watch object that can be used for asynchronous RTSP operations.


GstRtsp.RTSPWatch

Opaque RTSP watch object that can be used for asynchronous RTSP operations.


GstRtsp.RTSPWatch

Opaque RTSP watch object that can be used for asynchronous RTSP operations.


Methods

gst_rtsp_watch_attach

guint
gst_rtsp_watch_attach (GstRTSPWatch * watch,
                       GMainContext * context)

Adds a GstRTSPWatch to a context so that it will be executed within that context.

Parameters:

watch

a GstRTSPWatch

context

a GMainContext (if NULL, the default context will be used)

Returns

the ID (greater than 0) for the watch within the GMainContext.


GstRtsp.RTSPWatch.prototype.attach

function GstRtsp.RTSPWatch.prototype.attach(context: GLib.MainContext): {
    // javascript wrapper for 'gst_rtsp_watch_attach'
}

Adds a GstRtsp.RTSPWatch to a context so that it will be executed within that context.

Parameters:

context (GLib.MainContext)

a GMainContext (if NULL, the default context will be used)

Returns (Number)

the ID (greater than 0) for the watch within the GMainContext.


GstRtsp.RTSPWatch.attach

def GstRtsp.RTSPWatch.attach (self, context):
    #python wrapper for 'gst_rtsp_watch_attach'

Adds a GstRtsp.RTSPWatch to a context so that it will be executed within that context.

Parameters:

context (GLib.MainContext)

a GMainContext (if NULL, the default context will be used)

Returns (int)

the ID (greater than 0) for the watch within the GMainContext.


gst_rtsp_watch_get_send_backlog

gst_rtsp_watch_get_send_backlog (GstRTSPWatch * watch,
                                 gsize * bytes,
                                 guint * messages)

Get the maximum amount of bytes and messages that will be queued in watch. See gst_rtsp_watch_set_send_backlog.

Parameters:

watch

a GstRTSPWatch

bytes ( [out][allow-none])

maximum bytes

messages ( [out][allow-none])

maximum messages

Since : 1.2


GstRtsp.RTSPWatch.prototype.get_send_backlog

function GstRtsp.RTSPWatch.prototype.get_send_backlog(): {
    // javascript wrapper for 'gst_rtsp_watch_get_send_backlog'
}

Get the maximum amount of bytes and messages that will be queued in watch. See GstRtsp.RTSPWatch.prototype.set_send_backlog.

Parameters:

Since : 1.2


GstRtsp.RTSPWatch.get_send_backlog

def GstRtsp.RTSPWatch.get_send_backlog (self):
    #python wrapper for 'gst_rtsp_watch_get_send_backlog'

Get the maximum amount of bytes and messages that will be queued in watch. See GstRtsp.RTSPWatch.set_send_backlog.

Parameters:

Since : 1.2


gst_rtsp_watch_reset

gst_rtsp_watch_reset (GstRTSPWatch * watch)

Reset watch, this is usually called after gst_rtsp_connection_do_tunnel when the file descriptors of the connection might have changed.

Parameters:

watch

a GstRTSPWatch


GstRtsp.RTSPWatch.prototype.reset

function GstRtsp.RTSPWatch.prototype.reset(): {
    // javascript wrapper for 'gst_rtsp_watch_reset'
}

Reset watch, this is usually called after GstRtsp.RTSPConnection.prototype.do_tunnel when the file descriptors of the connection might have changed.

Parameters:


GstRtsp.RTSPWatch.reset

def GstRtsp.RTSPWatch.reset (self):
    #python wrapper for 'gst_rtsp_watch_reset'

Reset watch, this is usually called after GstRtsp.RTSPConnection.do_tunnel when the file descriptors of the connection might have changed.

Parameters:


gst_rtsp_watch_send_message

GstRTSPResult
gst_rtsp_watch_send_message (GstRTSPWatch * watch,
                             GstRTSPMessage * message,
                             guint * id)

Send a message using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

Parameters:

watch

a GstRTSPWatch

message

a GstRTSPMessage

id ( [out][allow-none])

location for a message ID or NULL

Returns

GST_RTSP_OK on success.


GstRtsp.RTSPWatch.prototype.send_message

function GstRtsp.RTSPWatch.prototype.send_message(message: GstRtsp.RTSPMessage): {
    // javascript wrapper for 'gst_rtsp_watch_send_message'
}

Send a message using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

Returns a tuple made of:

id (Number )

GstRtsp.RTSPResult.OK on success.


GstRtsp.RTSPWatch.send_message

def GstRtsp.RTSPWatch.send_message (self, message):
    #python wrapper for 'gst_rtsp_watch_send_message'

Send a message using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

Returns a tuple made of:

id (int )

GstRtsp.RTSPResult.OK on success.


gst_rtsp_watch_send_messages

GstRTSPResult
gst_rtsp_watch_send_messages (GstRTSPWatch * watch,
                              GstRTSPMessage * messages,
                              guint n_messages,
                              guint * id)

Sends messages using the connection of the watch. If they cannot be sent immediately, they will be queued for transmission in watch. The contents of messages will then be serialized and transmitted when the connection of the watch becomes writable. In case the messages are queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback once the last message is sent. The callback will only be called once for the last message.

Parameters:

watch

a GstRTSPWatch

messages ( [arraylength=n_messages])

the messages to send

n_messages

the number of messages to send

id ( [out][allow-none])

location for a message ID or NULL

Returns

GST_RTSP_OK on success.

Since : 1.16


GstRtsp.RTSPWatch.prototype.send_messages

function GstRtsp.RTSPWatch.prototype.send_messages(messages: [ GstRtsp.RTSPMessage ], n_messages: Number): {
    // javascript wrapper for 'gst_rtsp_watch_send_messages'
}

Sends messages using the connection of the watch. If they cannot be sent immediately, they will be queued for transmission in watch. The contents of messages will then be serialized and transmitted when the connection of the watch becomes writable. In case the messages are queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback once the last message is sent. The callback will only be called once for the last message.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (Number)

the number of messages to send

Returns a tuple made of:

id (Number )

GstRtsp.RTSPResult.OK on success.

Since : 1.16


GstRtsp.RTSPWatch.send_messages

def GstRtsp.RTSPWatch.send_messages (self, messages, n_messages):
    #python wrapper for 'gst_rtsp_watch_send_messages'

Sends messages using the connection of the watch. If they cannot be sent immediately, they will be queued for transmission in watch. The contents of messages will then be serialized and transmitted when the connection of the watch becomes writable. In case the messages are queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback once the last message is sent. The callback will only be called once for the last message.

Parameters:

messages ([ GstRtsp.RTSPMessage ])

the messages to send

n_messages (int)

the number of messages to send

Returns a tuple made of:

id (int )

GstRtsp.RTSPResult.OK on success.

Since : 1.16


gst_rtsp_watch_set_flushing

gst_rtsp_watch_set_flushing (GstRTSPWatch * watch,
                             gboolean flushing)

When flushing is TRUE, abort a call to gst_rtsp_watch_wait_backlog and make sure gst_rtsp_watch_write_data returns immediately with GST_RTSP_EINTR. And empty the queue.

Parameters:

watch

a GstRTSPWatch

flushing

new flushing state

Since : 1.4


GstRtsp.RTSPWatch.prototype.set_flushing

function GstRtsp.RTSPWatch.prototype.set_flushing(flushing: Number): {
    // javascript wrapper for 'gst_rtsp_watch_set_flushing'
}

When flushing is true, abort a call to GstRtsp.RTSPWatch.prototype.wait_backlog and make sure GstRtsp.RTSPWatch.prototype.write_data returns immediately with GstRtsp.RTSPResult.EINTR. And empty the queue.

Parameters:

flushing (Number)

new flushing state

Since : 1.4


GstRtsp.RTSPWatch.set_flushing

def GstRtsp.RTSPWatch.set_flushing (self, flushing):
    #python wrapper for 'gst_rtsp_watch_set_flushing'

When flushing is True, abort a call to GstRtsp.RTSPWatch.wait_backlog and make sure GstRtsp.RTSPWatch.write_data returns immediately with GstRtsp.RTSPResult.EINTR. And empty the queue.

Parameters:

flushing (bool)

new flushing state

Since : 1.4


gst_rtsp_watch_set_send_backlog

gst_rtsp_watch_set_send_backlog (GstRTSPWatch * watch,
                                 gsize bytes,
                                 guint messages)

Set the maximum amount of bytes and messages that will be queued in watch. When the maximum amounts are exceeded, gst_rtsp_watch_write_data and gst_rtsp_watch_send_message will return GST_RTSP_ENOMEM.

A value of 0 for bytes or messages means no limits.

Parameters:

watch

a GstRTSPWatch

bytes

maximum bytes

messages

maximum messages

Since : 1.2


GstRtsp.RTSPWatch.prototype.set_send_backlog

function GstRtsp.RTSPWatch.prototype.set_send_backlog(bytes: Number, messages: Number): {
    // javascript wrapper for 'gst_rtsp_watch_set_send_backlog'
}

Set the maximum amount of bytes and messages that will be queued in watch. When the maximum amounts are exceeded, GstRtsp.RTSPWatch.prototype.write_data and GstRtsp.RTSPWatch.prototype.send_message will return GstRtsp.RTSPResult.ENOMEM.

A value of 0 for bytes or messages means no limits.

Parameters:

bytes (Number)

maximum bytes

messages (Number)

maximum messages

Since : 1.2


GstRtsp.RTSPWatch.set_send_backlog

def GstRtsp.RTSPWatch.set_send_backlog (self, bytes, messages):
    #python wrapper for 'gst_rtsp_watch_set_send_backlog'

Set the maximum amount of bytes and messages that will be queued in watch. When the maximum amounts are exceeded, GstRtsp.RTSPWatch.write_data and GstRtsp.RTSPWatch.send_message will return GstRtsp.RTSPResult.ENOMEM.

A value of 0 for bytes or messages means no limits.

Parameters:

bytes (int)

maximum bytes

messages (int)

maximum messages

Since : 1.2


gst_rtsp_watch_unref

gst_rtsp_watch_unref (GstRTSPWatch * watch)

Decreases the reference count of watch by one. If the resulting reference count is zero the watch and associated memory will be destroyed.

Parameters:

watch

a GstRTSPWatch


GstRtsp.RTSPWatch.prototype.unref

function GstRtsp.RTSPWatch.prototype.unref(): {
    // javascript wrapper for 'gst_rtsp_watch_unref'
}

Decreases the reference count of watch by one. If the resulting reference count is zero the watch and associated memory will be destroyed.

Parameters:


GstRtsp.RTSPWatch.unref

def GstRtsp.RTSPWatch.unref (self):
    #python wrapper for 'gst_rtsp_watch_unref'

Decreases the reference count of watch by one. If the resulting reference count is zero the watch and associated memory will be destroyed.

Parameters:


gst_rtsp_watch_wait_backlog

GstRTSPResult
gst_rtsp_watch_wait_backlog (GstRTSPWatch * watch,
                             GTimeVal * timeout)

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is NULL this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GST_RTSP_ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

watch

a GstRTSPWatch

timeout

a GTimeVal timeout

Returns

GST_RTSP_OK when if there is room in queue. GST_RTSP_ETIMEOUT when timeout was reached. GST_RTSP_EINTR when watch is flushing GST_RTSP_EINVAL when called with invalid parameters.

Since : 1.4

deprecated : 1.18


GstRtsp.RTSPWatch.prototype.wait_backlog

function GstRtsp.RTSPWatch.prototype.wait_backlog(timeout: GLib.TimeVal): {
    // javascript wrapper for 'gst_rtsp_watch_wait_backlog'
}

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is null this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GstRtsp.RTSPResult.ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when if there is room in queue. GstRtsp.RTSPResult.ETIMEOUT when timeout was reached. GstRtsp.RTSPResult.EINTR when watch is flushing GstRtsp.RTSPResult.EINVAL when called with invalid parameters.

Since : 1.4

deprecated : 1.18


GstRtsp.RTSPWatch.wait_backlog

def GstRtsp.RTSPWatch.wait_backlog (self, timeout):
    #python wrapper for 'gst_rtsp_watch_wait_backlog'

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is None this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GstRtsp.RTSPResult.ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

timeout (GLib.TimeVal)

a GTimeVal timeout

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when if there is room in queue. GstRtsp.RTSPResult.ETIMEOUT when timeout was reached. GstRtsp.RTSPResult.EINTR when watch is flushing GstRtsp.RTSPResult.EINVAL when called with invalid parameters.

Since : 1.4

deprecated : 1.18


gst_rtsp_watch_wait_backlog_usec

GstRTSPResult
gst_rtsp_watch_wait_backlog_usec (GstRTSPWatch * watch,
                                  gint64 timeout)

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GST_RTSP_ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GST_RTSP_ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

watch

a GstRTSPWatch

timeout

a timeout in microseconds

Returns

GST_RTSP_OK when if there is room in queue. GST_RTSP_ETIMEOUT when timeout was reached. GST_RTSP_EINTR when watch is flushing GST_RTSP_EINVAL when called with invalid parameters.

Since : 1.18


GstRtsp.RTSPWatch.prototype.wait_backlog_usec

function GstRtsp.RTSPWatch.prototype.wait_backlog_usec(timeout: Number): {
    // javascript wrapper for 'gst_rtsp_watch_wait_backlog_usec'
}

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GstRtsp.RTSPResult.ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

timeout (Number)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when if there is room in queue. GstRtsp.RTSPResult.ETIMEOUT when timeout was reached. GstRtsp.RTSPResult.EINTR when watch is flushing GstRtsp.RTSPResult.EINVAL when called with invalid parameters.

Since : 1.18


GstRtsp.RTSPWatch.wait_backlog_usec

def GstRtsp.RTSPWatch.wait_backlog_usec (self, timeout):
    #python wrapper for 'gst_rtsp_watch_wait_backlog_usec'

Wait until there is place in the backlog queue, timeout is reached or watch is set to flushing.

If timeout is 0 this function can block forever. If timeout contains a valid timeout, this function will return GstRtsp.RTSPResult.ETIMEOUT after the timeout expired.

The typically use of this function is when gst_rtsp_watch_write_data returns GstRtsp.RTSPResult.ENOMEM. The caller then calls this function to wait for free space in the backlog queue and try again.

Parameters:

timeout (int)

a timeout in microseconds

Returns (GstRtsp.RTSPResult)

GstRtsp.RTSPResult.OK when if there is room in queue. GstRtsp.RTSPResult.ETIMEOUT when timeout was reached. GstRtsp.RTSPResult.EINTR when watch is flushing GstRtsp.RTSPResult.EINVAL when called with invalid parameters.

Since : 1.18


gst_rtsp_watch_write_data

GstRTSPResult
gst_rtsp_watch_write_data (GstRTSPWatch * watch,
                           const guint8 * data,
                           guint size,
                           guint * id)

Write data using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

This function will take ownership of data and g_free it after use.

If the amount of queued data exceeds the limits set with gst_rtsp_watch_set_send_backlog, this function will return GST_RTSP_ENOMEM.

Parameters:

watch

a GstRTSPWatch

data ( [arraylength=size][transfer: full])

the data to queue

size

the size of data

id ( [out][allow-none])

location for a message ID or NULL

Returns

GST_RTSP_OK on success. GST_RTSP_ENOMEM when the backlog limits are reached. GST_RTSP_EINTR when watch was flushing.


GstRtsp.RTSPWatch.prototype.write_data

function GstRtsp.RTSPWatch.prototype.write_data(data: [ Number ], size: Number): {
    // javascript wrapper for 'gst_rtsp_watch_write_data'
}

Write data using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

This function will take ownership of data and GLib.prototype.free it after use.

If the amount of queued data exceeds the limits set with GstRtsp.RTSPWatch.prototype.set_send_backlog, this function will return GstRtsp.RTSPResult.ENOMEM.

Parameters:

data ([ Number ])

the data to queue

size (Number)

the size of data

Returns a tuple made of:

GstRtsp.RTSPResult.OK on success. GstRtsp.RTSPResult.ENOMEM when the backlog limits are reached. GstRtsp.RTSPResult.EINTR when watch was flushing.

id (Number )

GstRtsp.RTSPResult.OK on success. GstRtsp.RTSPResult.ENOMEM when the backlog limits are reached. GstRtsp.RTSPResult.EINTR when watch was flushing.


GstRtsp.RTSPWatch.write_data

def GstRtsp.RTSPWatch.write_data (self, data, size):
    #python wrapper for 'gst_rtsp_watch_write_data'

Write data using the connection of the watch. If it cannot be sent immediately, it will be queued for transmission in watch. The contents of message will then be serialized and transmitted when the connection of the watch becomes writable. In case the message is queued, the ID returned in id will be non-zero and used as the ID argument in the message_sent callback.

This function will take ownership of data and GLib.free it after use.

If the amount of queued data exceeds the limits set with GstRtsp.RTSPWatch.set_send_backlog, this function will return GstRtsp.RTSPResult.ENOMEM.

Parameters:

data ([ int ])

the data to queue

size (int)

the size of data

Returns a tuple made of:

GstRtsp.RTSPResult.OK on success. GstRtsp.RTSPResult.ENOMEM when the backlog limits are reached. GstRtsp.RTSPResult.EINTR when watch was flushing.

id (int )

GstRtsp.RTSPResult.OK on success. GstRtsp.RTSPResult.ENOMEM when the backlog limits are reached. GstRtsp.RTSPResult.EINTR when watch was flushing.


Functions

gst_rtsp_watch_new

GstRTSPWatch *
gst_rtsp_watch_new (GstRTSPConnection * conn,
                    GstRTSPWatchFuncs * funcs,
                    gpointer user_data,
                    GDestroyNotify notify)

Create a watch object for conn. The functions provided in funcs will be called with user_data when activity happened on the watch.

The new watch is usually created so that it can be attached to a maincontext with gst_rtsp_watch_attach.

conn must exist for the entire lifetime of the watch.

Parameters:

conn

a GstRTSPConnection

funcs

watch functions

user_data

user data to pass to funcs

notify

notify when user_data is not referenced anymore

Returns

a GstRTSPWatch that can be used for asynchronous RTSP communication. Free with gst_rtsp_watch_unref () after usage.


GstRTSPWatchFuncs

Callback functions from a GstRTSPWatch.


GstRtsp.RTSPWatchFuncs

Callback functions from a GstRtsp.RTSPWatch.


GstRtsp.RTSPWatchFuncs

Callback functions from a GstRtsp.RTSPWatch.


Callbacks

GstRTSPConnectionAcceptCertificateFunc

gboolean
(*GstRTSPConnectionAcceptCertificateFunc) (GTlsConnection * conn,
                                           GTlsCertificate * peer_cert,
                                           GTlsCertificateFlags errors,
                                           gpointer user_data)

Parameters:

conn
No description available
peer_cert
No description available
errors
No description available
user_data
No description available
Returns
No description available

GstRtsp.RTSPConnectionAcceptCertificateFunc

function GstRtsp.RTSPConnectionAcceptCertificateFunc(conn: Gio.TlsConnection, peer_cert: Gio.TlsCertificate, errors: Gio.TlsCertificateFlags, user_data: Object): {
    // javascript wrapper for 'GstRTSPConnectionAcceptCertificateFunc'
}

Parameters:

conn (Gio.TlsConnection)
No description available
peer_cert (Gio.TlsCertificate)
No description available
No description available
user_data (Object)
No description available
Returns (Number)
No description available

GstRtsp.RTSPConnectionAcceptCertificateFunc

def GstRtsp.RTSPConnectionAcceptCertificateFunc (conn, peer_cert, errors, *user_data):
    #python wrapper for 'GstRTSPConnectionAcceptCertificateFunc'

Parameters:

conn (Gio.TlsConnection)
No description available
peer_cert (Gio.TlsCertificate)
No description available
No description available
user_data (variadic)
No description available
Returns (bool)
No description available

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