rtsp client

GstRtspServer.RTSPClient

GObject.Object
    ╰──GstRtspServer.RTSPClient
        ╰──GstRtspServer.RTSPOnvifClient

GstRtspServer.RTSPClient.prototype.new

function GstRtspServer.RTSPClient.prototype.new(): {
    // javascript wrapper for 'gst_rtsp_client_new'
}

Create a new GstRtspServer.RTSPClient instance.

GstRtspServer.RTSPClient.prototype.attach

function GstRtspServer.RTSPClient.prototype.attach(context: GMainContext (not introspectable)): {
    // javascript wrapper for 'gst_rtsp_client_attach'
}

Attaches client to context. When the mainloop for context is run, the client will be dispatched. When context is null, the default context will be used).

This function should be called when the client properties and urls are fully configured and the client is ready to start.

GstRtspServer.RTSPClient.prototype.close

function GstRtspServer.RTSPClient.prototype.close(): {
    // javascript wrapper for 'gst_rtsp_client_close'
}

Close the connection of client and remove all media it was managing.

Since : 1.4

GstRtspServer.RTSPClient.prototype.get_auth

function GstRtspServer.RTSPClient.prototype.get_auth(): {
    // javascript wrapper for 'gst_rtsp_client_get_auth'
}

Get the GstRtspServer.RTSPAuth used as the authentication manager of client.

GstRtspServer.RTSPClient.prototype.get_connection

function GstRtspServer.RTSPClient.prototype.get_connection(): {
    // javascript wrapper for 'gst_rtsp_client_get_connection'
}

Get the GstRtsp.RTSPConnection of client.

GstRtspServer.RTSPClient.prototype.get_content_length_limit

function GstRtspServer.RTSPClient.prototype.get_content_length_limit(): {
    // javascript wrapper for 'gst_rtsp_client_get_content_length_limit'
}

Get the Content-Length limit of client.

Since : 1.18

GstRtspServer.RTSPClient.prototype.get_mount_points

function GstRtspServer.RTSPClient.prototype.get_mount_points(): {
    // javascript wrapper for 'gst_rtsp_client_get_mount_points'
}

Get the GstRtspServer.RTSPMountPoints object that client uses to manage its sessions.

GstRtspServer.RTSPClient.prototype.get_session_pool

function GstRtspServer.RTSPClient.prototype.get_session_pool(): {
    // javascript wrapper for 'gst_rtsp_client_get_session_pool'
}

Get the GstRtspServer.RTSPSessionPool object that client uses to manage its sessions.

GstRtspServer.RTSPClient.prototype.get_stream_transport

function GstRtspServer.RTSPClient.prototype.get_stream_transport(channel: Number): {
    // javascript wrapper for 'gst_rtsp_client_get_stream_transport'
}

This is useful when providing a send function through GstRtspServer.RTSPClient.prototype.set_send_func when doing RTSP over TCP: the send function must call gst_rtsp_stream_transport_message_sent () on the appropriate transport when data has been received for streaming to continue.

Since : 1.18

GstRtspServer.RTSPClient.prototype.get_thread_pool

function GstRtspServer.RTSPClient.prototype.get_thread_pool(): {
    // javascript wrapper for 'gst_rtsp_client_get_thread_pool'
}

Get the GstRtspServer.RTSPThreadPool used as the thread pool of client.

GstRtspServer.RTSPClient.prototype.handle_message

function GstRtspServer.RTSPClient.prototype.handle_message(message: GstRtsp.RTSPMessage): {
    // javascript wrapper for 'gst_rtsp_client_handle_message'
}

Let the client handle message.

GstRtspServer.RTSPClient.prototype.send_message

function GstRtspServer.RTSPClient.prototype.send_message(session: GstRtspServer.RTSPSession, message: GstRtsp.RTSPMessage): {
    // javascript wrapper for 'gst_rtsp_client_send_message'
}

Send a message message to the remote end. message must be a GstRtsp.RTSPMsgType.REQUEST or a GstRtsp.RTSPMsgType.RESPONSE.

GstRtspServer.RTSPClient.prototype.session_filter

function GstRtspServer.RTSPClient.prototype.session_filter(func: GstRtspServer.RTSPClientSessionFilterFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_client_session_filter'
}

Call func for each session managed by client. The result value of func determines what happens to the session. func will be called with client locked so no further actions on client can be performed from func.

If func returns GstRtspServer.RTSPFilterResult.REMOVE, the session will be removed from client.

If func returns GstRtspServer.RTSPFilterResult.KEEP, the session will remain in client.

If func returns GstRtspServer.RTSPFilterResult.REF, the session will remain in client but will also be added with an additional ref to the result GLib.List of this function..

When func is null, GstRtspServer.RTSPFilterResult.REF will be assumed for each session.

GstRtspServer.RTSPClient.prototype.set_auth

function GstRtspServer.RTSPClient.prototype.set_auth(auth: GstRtspServer.RTSPAuth): {
    // javascript wrapper for 'gst_rtsp_client_set_auth'
}

configure auth to be used as the authentication manager of client.

GstRtspServer.RTSPClient.prototype.set_connection

function GstRtspServer.RTSPClient.prototype.set_connection(conn: GstRtsp.RTSPConnection): {
    // javascript wrapper for 'gst_rtsp_client_set_connection'
}

Set the GstRtsp.RTSPConnection of client. This function takes ownership of conn.

GstRtspServer.RTSPClient.prototype.set_content_length_limit

function GstRtspServer.RTSPClient.prototype.set_content_length_limit(limit: Number): {
    // javascript wrapper for 'gst_rtsp_client_set_content_length_limit'
}

Configure client to use the specified Content-Length limit.

Define an appropriate request size limit and reject requests exceeding the limit with response status 413 Request Entity Too Large

Since : 1.18

GstRtspServer.RTSPClient.prototype.set_mount_points

function GstRtspServer.RTSPClient.prototype.set_mount_points(mounts: GstRtspServer.RTSPMountPoints): {
    // javascript wrapper for 'gst_rtsp_client_set_mount_points'
}

Set mounts as the mount points for client which it will use to map urls to media streams. These mount points are usually inherited from the server that created the client but can be overriden later.

GstRtspServer.RTSPClient.prototype.set_send_func

function GstRtspServer.RTSPClient.prototype.set_send_func(func: GstRtspServer.RTSPClientSendFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_client_set_send_func'
}

Set func as the callback that will be called when a new message needs to be sent to the client. user_data is passed to func and notify is called when user_data is no longer in use.

By default, the client will send the messages on the GstRtsp.RTSPConnection that was configured with GstRtspServer.RTSPClient.prototype.attach was called.

It is only allowed to set either a send_func or a send_messages_func but not both at the same time.

GstRtspServer.RTSPClient.prototype.set_send_messages_func

function GstRtspServer.RTSPClient.prototype.set_send_messages_func(func: GstRtspServer.RTSPClientSendMessagesFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_client_set_send_messages_func'
}

Set func as the callback that will be called when new messages needs to be sent to the client. user_data is passed to func and notify is called when user_data is no longer in use.

By default, the client will send the messages on the GstRtsp.RTSPConnection that was configured with GstRtspServer.RTSPClient.prototype.attach was called.

It is only allowed to set either a send_func or a send_messages_func but not both at the same time.

Since : 1.16

GstRtspServer.RTSPClient.prototype.set_session_pool

function GstRtspServer.RTSPClient.prototype.set_session_pool(pool: GstRtspServer.RTSPSessionPool): {
    // javascript wrapper for 'gst_rtsp_client_set_session_pool'
}

Set pool as the sessionpool for client which it will use to find or allocate sessions. the sessionpool is usually inherited from the server that created the client but can be overridden later.

GstRtspServer.RTSPClient.prototype.set_thread_pool

function GstRtspServer.RTSPClient.prototype.set_thread_pool(pool: GstRtspServer.RTSPThreadPool): {
    // javascript wrapper for 'gst_rtsp_client_set_thread_pool'
}

configure pool to be used as the thread pool of client.

announce-request

function announce_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'announce-request' signal
}

Flags: Run Last

check-requirements

function check_requirements_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, arr: [ String ], user_data: Object): {
    // javascript callback for the 'check-requirements' signal
}

Flags: Run Last

Since : 1.6

closed

function closed_callback(self: GstRtspServer.RTSPClient, user_data: Object): {
    // javascript callback for the 'closed' signal
}

Flags: Run Last

describe-request

function describe_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'describe-request' signal
}

Flags: Run Last

get-parameter-request

function get_parameter_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'get-parameter-request' signal
}

Flags: Run Last

handle-response

function handle_response_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'handle-response' signal
}

Flags: Run Last

new-session

function new_session_callback(self: GstRtspServer.RTSPClient, object: GstRtspServer.RTSPSession, user_data: Object): {
    // javascript callback for the 'new-session' signal
}

Flags: Run Last

options-request

function options_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'options-request' signal
}

Flags: Run Last

pause-request

function pause_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pause-request' signal
}

Flags: Run Last

play-request

function play_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'play-request' signal
}

Flags: Run Last

pre-announce-request

function pre_announce_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-announce-request' signal
}

Flags: Run Last

Since : 1.12

pre-describe-request

function pre_describe_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-describe-request' signal
}

Flags: Run Last

Since : 1.12

pre-get-parameter-request

function pre_get_parameter_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-get-parameter-request' signal
}

Flags: Run Last

Since : 1.12

pre-options-request

function pre_options_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-options-request' signal
}

Flags: Run Last

Since : 1.12

pre-pause-request

function pre_pause_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-pause-request' signal
}

Flags: Run Last

Since : 1.12

pre-play-request

function pre_play_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-play-request' signal
}

Flags: Run Last

Since : 1.12

pre-record-request

function pre_record_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-record-request' signal
}

Flags: Run Last

Since : 1.12

pre-set-parameter-request

function pre_set_parameter_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-set-parameter-request' signal
}

Flags: Run Last

Since : 1.12

pre-setup-request

function pre_setup_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-setup-request' signal
}

Flags: Run Last

Since : 1.12

pre-teardown-request

function pre_teardown_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'pre-teardown-request' signal
}

Flags: Run Last

Since : 1.12

record-request

function record_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'record-request' signal
}

Flags: Run Last

send-message

function send_message_callback(self: GstRtspServer.RTSPClient, session: GstRtspServer.RTSPSession, message: GstRtsp.RTSPMessage, user_data: Object): {
    // javascript callback for the 'send-message' signal
}

Flags: Run Last

set-parameter-request

function set_parameter_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'set-parameter-request' signal
}

Flags: Run Last

setup-request

function setup_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'setup-request' signal
}

Flags: Run Last

teardown-request

function teardown_request_callback(self: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, user_data: Object): {
    // javascript callback for the 'teardown-request' signal
}

Flags: Run Last

drop-backlog

“drop-backlog” Number

Flags : Read / Write

mount-points

“mount-points” GstRtspServer.RTSPMountPoints

Flags : Read / Write

post-session-timeout

“post-session-timeout” Number

Flags : Read / Write

session-pool

“session-pool” GstRtspServer.RTSPSessionPool

Flags : Read / Write

vfunc_adjust_error_code

function vfunc_adjust_error_code(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, code: GstRtsp.RTSPStatusCode): {
    // javascript implementation of the 'adjust_error_code' virtual method
}

Called before sending error response to give the application the possibility to adjust the error code.

Since : 1.22

vfunc_adjust_play_mode

function vfunc_adjust_play_mode(client: GstRtspServer.RTSPClient, context: GstRtspServer.RTSPContext, range: GstRtsp.RTSPTimeRange, flags: Gst.SeekFlags, rate: Number, trickmode_interval: Number, enable_rate_control: Number): {
    // javascript implementation of the 'adjust_play_mode' virtual method
}

called to give the application the possibility to adjust the range, seek flags, rate and rate-control. Since 1.18

vfunc_adjust_play_response

function vfunc_adjust_play_response(client: GstRtspServer.RTSPClient, context: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'adjust_play_response' virtual method
}

called to give the implementation the possibility to adjust the response to a play request, for example if extra headers were parsed when vfunc_adjust_play_mode was called. Since 1.18

vfunc_announce_request

function vfunc_announce_request(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'announce_request' virtual method
}

vfunc_check_requirements

function vfunc_check_requirements(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, arr: String): {
    // javascript implementation of the 'check_requirements' virtual method
}

vfunc_closed

function vfunc_closed(client: GstRtspServer.RTSPClient): {
    // javascript implementation of the 'closed' virtual method
}

vfunc_configure_client_media

function vfunc_configure_client_media(client: GstRtspServer.RTSPClient, media: GstRtspServer.RTSPMedia, stream: GstRtspServer.RTSPStream, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'configure_client_media' virtual method
}

called when the stream in media needs to be configured. The default implementation will configure the blocksize on the payloader when spcified in the request headers.

vfunc_configure_client_transport

function vfunc_configure_client_transport(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, ct: GstRtsp.RTSPTransport): {
    // javascript implementation of the 'configure_client_transport' virtual method
}

called when the client transport needs to be configured.

vfunc_create_sdp

function vfunc_create_sdp(client: GstRtspServer.RTSPClient, media: GstRtspServer.RTSPMedia): {
    // javascript implementation of the 'create_sdp' virtual method
}

called when the SDP needs to be created for media.

vfunc_describe_request

function vfunc_describe_request(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'describe_request' virtual method
}

vfunc_get_parameter_request

function vfunc_get_parameter_request(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'get_parameter_request' virtual method
}

vfunc_handle_response

function vfunc_handle_response(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'handle_response' virtual method
}

vfunc_handle_sdp

function vfunc_handle_sdp(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext, media: GstRtspServer.RTSPMedia, sdp: GstSdp.SDPMessage): {
    // javascript implementation of the 'handle_sdp' virtual method
}

vfunc_make_path_from_uri

function vfunc_make_path_from_uri(client: GstRtspServer.RTSPClient, uri: GstRtsp.RTSPUrl): {
    // javascript implementation of the 'make_path_from_uri' virtual method
}

called to create path from uri.

vfunc_new_session

function vfunc_new_session(client: GstRtspServer.RTSPClient, session: GstRtspServer.RTSPSession): {
    // javascript implementation of the 'new_session' virtual method
}

vfunc_options_request

function vfunc_options_request(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'options_request' virtual method
}

vfunc_params_get

function vfunc_params_get(client: GstRtspServer.RTSPClient, ctx: GstRtspServer.RTSPContext): {
    // javascript implementation of the 'params_get' virtual method
}

get parameters. This function should also initialize the RTSP response(ctx->response) via a call to GstRtsp.RTSPMessage.prototype.init_response