GstRTSPConnection

GstRtsp.RTSPConnection

GstRtsp.RTSPConnection.prototype.add_extra_http_request_header

function GstRtsp.RTSPConnection.prototype.add_extra_http_request_header(key: String, value: String): {
    // javascript wrapper for 'gst_rtsp_connection_add_extra_http_request_header'
}

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.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.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.

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.

deprecated : 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.

Since : 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.

Since : 1.8

deprecated : 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.

Since : 1.18

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.

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.

GstRtsp.RTSPConnection.prototype.free

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

Close and free conn.

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.

Since : 1.20

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.

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.

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'
}

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.

Since : 1.2

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.

Since : 1.4

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.

Since : 1.6

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.

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.prototype.get_tunnelid

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

Get the tunnel session id the connection.

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.

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.

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.

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.

deprecated : 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

Since : 1.18

GstRtsp.RTSPConnection.prototype.poll

function GstRtsp.RTSPConnection.prototype.poll(events: 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.

deprecated : 1.18

GstRtsp.RTSPConnection.prototype.poll_usec

function GstRtsp.RTSPConnection.prototype.poll_usec(events: 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.

Since : 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.

deprecated : 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.

Since : 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.

deprecated : 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.

Since : 1.18

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.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.

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.

Since : 1.16

deprecated : 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.

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.

Since : 1.18

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.

Since : 1.14

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.

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.

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.

Since : 1.18

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.

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.

Since : 1.20

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.

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.

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.

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

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.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.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.

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.

Since : 1.2.1

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.

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.

deprecated : 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.

Since : 1.18

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.

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.

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.

GstRtsp.RTSPWatch

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.

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.

Since : 1.2

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.

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.

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.

Since : 1.16

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.

Since : 1.4

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.

Since : 1.2

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.