GstRTSPConnection
This object manages the RTSP connection to the server. It provides function
to receive and send bytes and messages.
gst_rtsp_connection_add_extra_http_request_header (GstRTSPConnection * conn,
const gchar * key,
const gchar * value)
Add header to be appended to any HTTP request made by connection.
If the header already exists then the old header is replaced by the new header.
Only applicable in HTTP tunnel mode.
Parameters:
value
–
HTTP header value
Since : 1.24
GstRtsp.RTSPConnection.prototype.add_extra_http_request_header
function GstRtsp.RTSPConnection.prototype.add_extra_http_request_header(key: String, value: String): {
}
Add header to be appended to any HTTP request made by connection.
If the header already exists then the old header is replaced by the new header.
Only applicable in HTTP tunnel mode.
Since : 1.24
GstRtsp.RTSPConnection.add_extra_http_request_header
def GstRtsp.RTSPConnection.add_extra_http_request_header (self, key, value):
Add header to be appended to any HTTP request made by connection.
If the header already exists then the old header is replaced by the new header.
Only applicable in HTTP tunnel mode.
Since : 1.24
gst_rtsp_connection_clear_auth_params
gst_rtsp_connection_clear_auth_params (GstRTSPConnection * conn)
Clear the list of authentication directives stored in conn.
GstRtsp.RTSPConnection.prototype.clear_auth_params
function GstRtsp.RTSPConnection.prototype.clear_auth_params(): {
}
Clear the list of authentication directives stored in conn.
GstRtsp.RTSPConnection.clear_auth_params
def GstRtsp.RTSPConnection.clear_auth_params (self):
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.
GstRtsp.RTSPConnection.prototype.close
function GstRtsp.RTSPConnection.prototype.close(): {
}
Close the connected conn. After this call, the connection is in the same
state as when it was first created.
GstRtsp.RTSPConnection.close
def GstRtsp.RTSPConnection.close (self):
Close the connected conn. After this call, the connection is in the same
state as when it was first created.
GstRtsp.RTSPConnection.connect_usec
def GstRtsp.RTSPConnection.connect_usec (self, timeout):
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:
a timeout in microseconds
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:
timeout
–
a GTimeVal timeout
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): {
}
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.
Since : 1.8
deprecated : 1.18
GstRtsp.RTSPConnection.connect_with_response
def GstRtsp.RTSPConnection.connect_with_response (self, timeout, 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.
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:
timeout
–
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.connect_with_response_usec
function GstRtsp.RTSPConnection.prototype.connect_with_response_usec(timeout: Number, response: GstRtsp.RTSPMessage): {
}
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:
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.connect_with_response_usec
def GstRtsp.RTSPConnection.connect_with_response_usec (self, timeout, response):
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:
a timeout in microseconds
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.
Returns
–
return GST_RTSP_OK on success.
GstRtsp.RTSPConnection.prototype.do_tunnel
function GstRtsp.RTSPConnection.prototype.do_tunnel(conn2: GstRtsp.RTSPConnection): {
}
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.
return GST_RTSP_OK on success.
GstRtsp.RTSPConnection.do_tunnel
def GstRtsp.RTSPConnection.do_tunnel (self, 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
GstRtsp.RTSPConnection.free.
If conn2 is None then only the base64 decoding context will be setup for
conn.
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:
flush
–
start or stop the flush
GstRtsp.RTSPConnection.prototype.flush
function GstRtsp.RTSPConnection.prototype.flush(flush: Number): {
}
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.
GstRtsp.RTSPConnection.flush
def GstRtsp.RTSPConnection.flush (self, 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.
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.
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(): {
}
Get the ignore_x_server_reply value.
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):
Get the ignore_x_server_reply value.
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.
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(): {
}
Retrieve the IP address of the other end of conn.
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):
Retrieve the IP address of the other end of conn.
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.
Returns
(
[transfer: none][nullable])
–
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(): {
}
Get the file descriptor for reading.
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):
Get the file descriptor for reading.
the file descriptor used for reading or None on
error. The file descriptor remains valid until the connection is closed.
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:
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(): {
}
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.
the TLS connection for conn.
Since : 1.2
GstRtsp.RTSPConnection.get_tls
@raises(GLib.GError)
def GstRtsp.RTSPConnection.get_tls (self):
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.
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.
Returns
(
[transfer: full][nullable])
–
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(): {
}
Gets the anchor certificate authorities database that will be used
after a server certificate can't be verified with the default
certificate database.
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):
Gets the anchor certificate authorities database that will be used
after a server certificate can't be verified with the default
certificate database.
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.
Since : 1.6
GstRtsp.RTSPConnection.prototype.get_tls_interaction
function GstRtsp.RTSPConnection.prototype.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.
Since : 1.6
GstRtsp.RTSPConnection.get_tls_interaction
def GstRtsp.RTSPConnection.get_tls_interaction (self):
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.
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.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to ignore G_TLS_CERTIFICATE_EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
Returns
–
the validation flags.
Since : 1.2.1
GstRtsp.RTSPConnection.prototype.get_tls_validation_flags
function GstRtsp.RTSPConnection.prototype.get_tls_validation_flags(): {
}
Gets the TLS validation flags used to verify the peer certificate
when a TLS connection is established.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to ignore Gio.TlsCertificateFlags.EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
Since : 1.2.1
GstRtsp.RTSPConnection.get_tls_validation_flags
def GstRtsp.RTSPConnection.get_tls_validation_flags (self):
Gets the TLS validation flags used to verify the peer certificate
when a TLS connection is established.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to ignore Gio.TlsCertificateFlags.EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
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.
returns a non-empty string if conn is being tunneled over HTTP.
GstRtsp.RTSPConnection.prototype.get_tunnelid
function GstRtsp.RTSPConnection.prototype.get_tunnelid(): {
}
Get the tunnel session id the connection.
returns a non-empty string if conn is being tunneled over HTTP.
GstRtsp.RTSPConnection.get_tunnelid
def GstRtsp.RTSPConnection.get_tunnelid (self):
Get the tunnel session id the connection.
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.
Returns
–
The URL. This value remains valid until the
connection is freed.
GstRtsp.RTSPConnection.prototype.get_url
function GstRtsp.RTSPConnection.prototype.get_url(): {
}
Retrieve the URL of the other end of conn.
The URL. This value remains valid until the
connection is freed.
GstRtsp.RTSPConnection.get_url
def GstRtsp.RTSPConnection.get_url (self):
Retrieve the URL of the other end of conn.
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.
Returns
(
[transfer: none][nullable])
–
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(): {
}
Get the file descriptor for writing.
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):
Get the file descriptor for writing.
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.
Returns
–
if conn is using HTTP tunneling.
GstRtsp.RTSPConnection.prototype.is_tunneled
function GstRtsp.RTSPConnection.prototype.is_tunneled(): {
}
Get the tunneling state of the connection.
if conn is using HTTP tunneling.
GstRtsp.RTSPConnection.is_tunneled
def GstRtsp.RTSPConnection.is_tunneled (self):
Get the tunneling state of the connection.
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.
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.next_timeout
function GstRtsp.RTSPConnection.prototype.next_timeout(timeout: GLib.TimeVal): {
}
Calculate the next timeout for conn, storing the result in timeout.
deprecated : 1.18
GstRtsp.RTSPConnection.next_timeout
def GstRtsp.RTSPConnection.next_timeout (self, timeout):
Calculate the next timeout for conn, storing the result in 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
Returns
–
the next timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.next_timeout_usec
function GstRtsp.RTSPConnection.prototype.next_timeout_usec(): {
}
Calculate the next timeout for conn
the next timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.next_timeout_usec
def GstRtsp.RTSPConnection.next_timeout_usec (self):
Calculate the next timeout for conn
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:
location for result flags
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.poll
function GstRtsp.RTSPConnection.prototype.poll(events: GstRtsp.RTSPEvent, timeout: GLib.TimeVal): {
}
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.
deprecated : 1.18
GstRtsp.RTSPConnection.poll
def GstRtsp.RTSPConnection.poll (self, events, 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 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.
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:
location for result flags
timeout
–
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.poll_usec
function GstRtsp.RTSPConnection.prototype.poll_usec(events: GstRtsp.RTSPEvent, timeout: Number): {
}
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:
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPConnection.poll_usec
def GstRtsp.RTSPConnection.poll_usec (self, events, 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 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:
a timeout in microseconds
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:
data
(
[arraylength=size])
–
the data to read
timeout
–
a timeout value or NULL
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.read
function GstRtsp.RTSPConnection.prototype.read(data: [ Number ], size: Number, timeout: GLib.TimeVal): {
}
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.
deprecated : 1.18
GstRtsp.RTSPConnection.read
def GstRtsp.RTSPConnection.read (self, data, size, timeout):
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.
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:
data
(
[arraylength=size])
–
the data to read
timeout
–
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.read_usec
function GstRtsp.RTSPConnection.prototype.read_usec(data: [ Number ], size: Number, timeout: Number): {
}
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:
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.read_usec
def GstRtsp.RTSPConnection.read_usec (self, data, size, 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 GstRtsp.RTSPConnection.flush.
Parameters:
a timeout value in microseconds
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:
message
(
[transfer: none])
–
the message to read
timeout
–
a timeout value or NULL
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.receive
function GstRtsp.RTSPConnection.prototype.receive(message: GstRtsp.RTSPMessage, timeout: GLib.TimeVal): {
}
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.
deprecated : 1.18
GstRtsp.RTSPConnection.receive
def GstRtsp.RTSPConnection.receive (self, message, timeout):
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.
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:
message
(
[transfer: none])
–
the message to read
timeout
–
a timeout value or 0
Since : 1.18
GstRtsp.RTSPConnection.prototype.receive_usec
function GstRtsp.RTSPConnection.prototype.receive_usec(message: GstRtsp.RTSPMessage, timeout: Number): {
}
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.
Since : 1.18
GstRtsp.RTSPConnection.receive_usec
def GstRtsp.RTSPConnection.receive_usec (self, message, 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 GstRtsp.RTSPConnection.flush.
Since : 1.18
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:
message
–
the message to send
timeout
–
a timeout value or NULL
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.send
function GstRtsp.RTSPConnection.prototype.send(message: GstRtsp.RTSPMessage, timeout: GLib.TimeVal): {
}
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.
deprecated : 1.18
GstRtsp.RTSPConnection.send
def GstRtsp.RTSPConnection.send (self, message, timeout):
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.
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:
messages
(
[arraylength=n_messages])
–
the messages to send
n_messages
–
the number of messages to send
timeout
–
a timeout value or NULL
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): {
}
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:
the number of messages to send
Since : 1.16
deprecated : 1.18
GstRtsp.RTSPConnection.send_messages
def GstRtsp.RTSPConnection.send_messages (self, messages, n_messages, timeout):
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:
the number of messages to send
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:
messages
(
[arraylength=n_messages])
–
the messages to send
n_messages
–
the number of messages to send
timeout
–
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.send_messages_usec
function GstRtsp.RTSPConnection.prototype.send_messages_usec(messages: [ GstRtsp.RTSPMessage ], n_messages: Number, timeout: Number): {
}
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:
the number of messages to send
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.send_messages_usec
def GstRtsp.RTSPConnection.send_messages_usec (self, messages, n_messages, 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 GstRtsp.RTSPConnection.flush.
Parameters:
the number of messages to send
a timeout value in microseconds
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:
message
–
the message to send
timeout
–
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.send_usec
function GstRtsp.RTSPConnection.prototype.send_usec(message: GstRtsp.RTSPMessage, timeout: Number): {
}
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:
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.send_usec
def GstRtsp.RTSPConnection.send_usec (self, message, 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 GstRtsp.RTSPConnection.flush.
Parameters:
a timeout value in microseconds
Since : 1.18
GstRtsp.RTSPConnection.prototype.set_accept_certificate_func
function GstRtsp.RTSPConnection.prototype.set_accept_certificate_func(func: GstRtsp.RTSPConnectionAcceptCertificateFunc, user_data: Object): {
}
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.
Since : 1.14
GstRtsp.RTSPConnection.set_accept_certificate_func
def GstRtsp.RTSPConnection.set_accept_certificate_func (self, func, *user_data):
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.
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:
method
–
authentication method
GstRtsp.RTSPConnection.prototype.set_auth
function GstRtsp.RTSPConnection.prototype.set_auth(method: GstRtsp.RTSPAuthMethod, user: String, pass: String): {
}
Configure conn for authentication mode method with user and pass as the
user and password respectively.
GstRtsp.RTSPConnection.set_auth
def GstRtsp.RTSPConnection.set_auth (self, method, user, pass):
Configure conn for authentication mode method with user and pass as the
user and password respectively.
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:
param
–
authentication directive
GstRtsp.RTSPConnection.prototype.set_auth_param
function GstRtsp.RTSPConnection.prototype.set_auth_param(param: String, value: String): {
}
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.
GstRtsp.RTSPConnection.set_auth_param
def GstRtsp.RTSPConnection.set_auth_param (self, param, value):
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.
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:
limit
–
Content-Length limit
Since : 1.18
GstRtsp.RTSPConnection.prototype.set_content_length_limit
function GstRtsp.RTSPConnection.prototype.set_content_length_limit(limit: Number): {
}
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.
Since : 1.18
GstRtsp.RTSPConnection.set_content_length_limit
def GstRtsp.RTSPConnection.set_content_length_limit (self, 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.
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:
enable
–
TRUE to enable manual HTTP mode
GstRtsp.RTSPConnection.prototype.set_http_mode
function GstRtsp.RTSPConnection.prototype.set_http_mode(enable: Number): {
}
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:
true to enable manual HTTP mode
GstRtsp.RTSPConnection.set_http_mode
def GstRtsp.RTSPConnection.set_http_mode (self, 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:
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:
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): {
}
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:
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):
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:
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.
GstRtsp.RTSPConnection.prototype.set_ip
function GstRtsp.RTSPConnection.prototype.set_ip(ip: String): {
}
Set the IP address of the server.
GstRtsp.RTSPConnection.set_ip
def GstRtsp.RTSPConnection.set_ip (self, ip):
Set the IP address of the server.
GstRtsp.RTSPConnection.prototype.set_proxy
function GstRtsp.RTSPConnection.prototype.set_proxy(host: String, port: Number): {
}
Set the proxy host and port.
GstRtsp.RTSPConnection.set_proxy
def GstRtsp.RTSPConnection.set_proxy (self, host, port):
Set the proxy host and port.
GstRtsp.RTSPConnection.prototype.set_qos_dscp
function GstRtsp.RTSPConnection.prototype.set_qos_dscp(qos_dscp: Number): {
}
Configure conn to use the specified DSCP value.
GstRtsp.RTSPConnection.set_qos_dscp
def GstRtsp.RTSPConnection.set_qos_dscp (self, qos_dscp):
Configure conn to use the specified DSCP value.
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:
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): {
}
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:
true if the connection should remember the session id
GstRtsp.RTSPConnection.set_remember_session_id
def GstRtsp.RTSPConnection.set_remember_session_id (self, remember):
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:
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.
Since : 1.4
GstRtsp.RTSPConnection.prototype.set_tls_database
function GstRtsp.RTSPConnection.prototype.set_tls_database(database: Gio.TlsDatabase): {
}
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):
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.
Since : 1.6
GstRtsp.RTSPConnection.prototype.set_tls_interaction
function GstRtsp.RTSPConnection.prototype.set_tls_interaction(interaction: Gio.TlsInteraction): {
}
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):
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.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to mask G_TLS_CERTIFICATE_EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
Parameters:
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): {
}
Sets the TLS validation flags to be used to verify the peer
certificate when a TLS connection is established.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to mask Gio.TlsCertificateFlags.EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
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):
Sets the TLS validation flags to be used to verify the peer
certificate when a TLS connection is established.
GLib guarantees that if certificate verification fails, at least one error
will be set, but it does not guarantee that all possible errors will be
set. Accordingly, you may not safely decide to ignore any particular type
of error.
For example, it would be incorrect to mask Gio.TlsCertificateFlags.EXPIRED if
you want to allow expired certificates, because this could potentially be
the only error flag set even if other problems exist with the certificate.
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.
GstRtsp.RTSPConnection.prototype.set_tunneled
function GstRtsp.RTSPConnection.prototype.set_tunneled(tunneled: Number): {
}
Set the HTTP tunneling state of the connection. This must be configured before
the conn is connected.
GstRtsp.RTSPConnection.set_tunneled
def GstRtsp.RTSPConnection.set_tunneled (self, tunneled):
Set the HTTP tunneling state of the connection. This must be configured before
the conn is connected.
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:
data
(
[arraylength=size])
–
the data to write
timeout
–
a timeout value or NULL
deprecated : 1.18
GstRtsp.RTSPConnection.prototype.write
function GstRtsp.RTSPConnection.prototype.write(data: [ Number ], size: Number, timeout: GLib.TimeVal): {
}
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.
deprecated : 1.18
GstRtsp.RTSPConnection.write
def GstRtsp.RTSPConnection.write (self, data, size, timeout):
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.
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:
data
(
[arraylength=size])
–
the data to write
timeout
–
a timeout value or 0
Since : 1.18
GstRtsp.RTSPConnection.prototype.write_usec
function GstRtsp.RTSPConnection.prototype.write_usec(data: [ Number ], size: Number, timeout: Number): {
}
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.
Since : 1.18
GstRtsp.RTSPConnection.write_usec
def GstRtsp.RTSPConnection.write_usec (self, data, size, 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 GstRtsp.RTSPConnection.flush.
Since : 1.18
GstRtsp.prototype.rtsp_connection_accept
function GstRtsp.prototype.rtsp_connection_accept(socket: Gio.Socket, cancellable: Gio.Cancellable): {
}
Accept a new connection on socket and create a new GstRtsp.RTSPConnection for
handling communication on new socket.
GstRtsp.rtsp_connection_accept
def GstRtsp.rtsp_connection_accept (socket, cancellable):
Accept a new connection on socket and create a new GstRtsp.RTSPConnection for
handling communication on new socket.
GstRtsp.prototype.rtsp_connection_create
function GstRtsp.prototype.rtsp_connection_create(url: GstRtsp.RTSPUrl): {
}
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.
GstRtsp.rtsp_connection_create
def GstRtsp.rtsp_connection_create (url):
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.
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:
ip
–
the IP address of the other end
port
–
the port used by the other end
initial_buffer
–
data already read from fd
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): {
}
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:
the IP address of the other end
the port used by the other end
data already read from fd
GstRtsp.rtsp_connection_create_from_socket
def GstRtsp.rtsp_connection_create_from_socket (socket, ip, port, initial_buffer):
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:
the IP address of the other end
the port used by the other end
data already read from fd
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:
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: GMainContext (not introspectable)): {
}
Adds a GstRtsp.RTSPWatch to a context so that it will be executed within that context.
Parameters:
a GMainContext (if NULL, the default context will be used)
the ID (greater than 0) for the watch within the GMainContext.
GstRtsp.RTSPWatch.attach
def GstRtsp.RTSPWatch.attach (self, context):
Adds a GstRtsp.RTSPWatch to a context so that it will be executed within that context.
Parameters:
a GMainContext (if NULL, the default context will be used)
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:
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(): {
}
Get the maximum amount of bytes and messages that will be queued in watch.
See GstRtsp.RTSPWatch.prototype.set_send_backlog.
Since : 1.2
GstRtsp.RTSPWatch.get_send_backlog
def GstRtsp.RTSPWatch.get_send_backlog (self):
Get the maximum amount of bytes and messages that will be queued in watch.
See GstRtsp.RTSPWatch.set_send_backlog.
Since : 1.2
GstRtsp.RTSPWatch.prototype.reset
function GstRtsp.RTSPWatch.prototype.reset(): {
}
Reset watch, this is usually called after GstRtsp.RTSPConnection.prototype.do_tunnel
when the file descriptors of the connection might have changed.
GstRtsp.RTSPWatch.reset
def GstRtsp.RTSPWatch.reset (self):
Reset watch, this is usually called after GstRtsp.RTSPConnection.do_tunnel
when the file descriptors of the connection might have changed.
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:
location for a message ID or NULL
GstRtsp.RTSPWatch.prototype.send_message
function GstRtsp.RTSPWatch.prototype.send_message(message: GstRtsp.RTSPMessage): {
}
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.
GstRtsp.RTSPWatch.send_message
def GstRtsp.RTSPWatch.send_message (self, 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.
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:
messages
(
[arraylength=n_messages])
–
the messages to send
n_messages
–
the number of messages to send
location for a message ID or NULL
Since : 1.16
GstRtsp.RTSPWatch.prototype.send_messages
function GstRtsp.RTSPWatch.prototype.send_messages(messages: [ GstRtsp.RTSPMessage ], n_messages: Number): {
}
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:
the number of messages to send
Since : 1.16
GstRtsp.RTSPWatch.send_messages
def GstRtsp.RTSPWatch.send_messages (self, messages, n_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:
the number of messages to send
Since : 1.16
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.
GstRtsp.RTSPWatch.prototype.unref
function GstRtsp.RTSPWatch.prototype.unref(): {
}
Decreases the reference count of watch by one. If the resulting reference
count is zero the watch and associated memory will be destroyed.
GstRtsp.RTSPWatch.unref
def GstRtsp.RTSPWatch.unref (self):
Decreases the reference count of watch by one. If the resulting reference
count is zero the watch and associated memory will be destroyed.
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:
timeout
–
a GTimeVal timeout
Since : 1.4
deprecated : 1.18
GstRtsp.RTSPWatch.prototype.wait_backlog
function GstRtsp.RTSPWatch.prototype.wait_backlog(timeout: GLib.TimeVal): {
}
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.
Since : 1.4
deprecated : 1.18
GstRtsp.RTSPWatch.wait_backlog
def GstRtsp.RTSPWatch.wait_backlog (self, timeout):
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.
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:
timeout
–
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPWatch.prototype.wait_backlog_usec
function GstRtsp.RTSPWatch.prototype.wait_backlog_usec(timeout: Number): {
}
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:
a timeout in microseconds
Since : 1.18
GstRtsp.RTSPWatch.wait_backlog_usec
def GstRtsp.RTSPWatch.wait_backlog_usec (self, 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 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:
a timeout in microseconds
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:
data
(
[arraylength=size][transfer: full])
–
the data to queue
location for a message ID or NULL
GstRtsp.RTSPWatch.prototype.write_data
function GstRtsp.RTSPWatch.prototype.write_data(data: [ Number ], size: Number): {
}
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.
GstRtsp.RTSPWatch.write_data
def GstRtsp.RTSPWatch.write_data (self, data, size):
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.
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:
user_data
–
user data to pass to funcs
notify
–
notify when user_data is not referenced anymore
Returns
(
[transfer: full])
–
a GstRTSPWatch that can be used for asynchronous RTSP
communication. Free with gst_rtsp_watch_unref () after usage.
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): {
}
GstRtsp.RTSPConnectionAcceptCertificateFunc
def GstRtsp.RTSPConnectionAcceptCertificateFunc (conn, peer_cert, errors, *user_data):