GstRtspServer.RTSPClient¶

Subclasses:: GstRtspServer.RTSPOnvifClient

Methods¶

Inherited:: GObject.Object (37)
Structs:: GObject.ObjectClass (5)

class	`new` ()
	`attach` (context)
	`close` ()
	`get_auth` ()
	`get_connection` ()
	`get_content_length_limit` ()
	`get_mount_points` ()
	`get_session_pool` ()
	`get_stream_transport` (channel)
	`get_thread_pool` ()
	`handle_message` (message)
	`send_message` (session, message)
	`session_filter` (func, *user_data)
	`set_auth` (auth)
	`set_connection` (conn)
	`set_content_length_limit` (limit)
	`set_mount_points` (mounts)
	`set_send_func` (func, *user_data)
	`set_send_messages_func` (func, *user_data)
	`set_session_pool` (pool)
	`set_thread_pool` (pool)

Virtual Methods¶

Inherited:: GObject.Object (7)

	`do_adjust_error_code` (ctx, code)
	`do_adjust_play_mode` (context, range, flags, rate, trickmode_interval, enable_rate_control)
	`do_adjust_play_response` (context)
	`do_announce_request` (ctx)
	`do_check_requirements` (ctx, arr)
	`do_closed` ()
	`do_configure_client_media` (media, stream, ctx)
	`do_configure_client_transport` (ctx, ct)
	`do_create_sdp` (media)
	`do_describe_request` (ctx)
	`do_get_parameter_request` (ctx)
	`do_handle_response` (ctx)
	`do_handle_sdp` (ctx, media, sdp)
	`do_make_path_from_uri` (uri)
	`do_new_session` (session)
	`do_options_request` (ctx)
	`do_params_get` (ctx)
	`do_params_set` (ctx)
	`do_pause_request` (ctx)
	`do_play_request` (ctx)
	`do_pre_announce_request` (ctx)
	`do_pre_describe_request` (ctx)
	`do_pre_get_parameter_request` (ctx)
	`do_pre_options_request` (ctx)
	`do_pre_pause_request` (ctx)
	`do_pre_play_request` (ctx)
	`do_pre_record_request` (ctx)
	`do_pre_set_parameter_request` (ctx)
	`do_pre_setup_request` (ctx)
	`do_pre_teardown_request` (ctx)
	`do_record_request` (ctx)
	`do_send_message` (ctx, response)
	`do_set_parameter_request` (ctx)
	`do_setup_request` (ctx)
	`do_teardown_request` (ctx)
	`do_tunnel_http_response` (request, response)

Properties¶

Name	Type	Flags	Short Description
`drop-backlog`	`bool`	r/w	Drop `GstRtsp.data` when the backlog queue is full
`mount-points`	`GstRtspServer.RTSPMountPoints`	r/w	The mount points to use for client session
`post-session-timeout`	`int`	r/w	An extra TCP connection timeout after session timeout
`session-pool`	`GstRtspServer.RTSPSessionPool`	r/w	The session pool to use for client session

Signals¶

Inherited:: GObject.Object (1)

Name	Short Description
`announce-request`
`check-requirements`
`closed`
`describe-request`
`get-parameter-request`
`handle-response`
`new-session`
`options-request`
`pause-request`
`play-request`
`pre-announce-request`
`pre-describe-request`
`pre-get-parameter-request`
`pre-options-request`
`pre-pause-request`
`pre-play-request`
`pre-record-request`
`pre-set-parameter-request`
`pre-setup-request`
`pre-teardown-request`
`record-request`
`send-message`
`set-parameter-request`
`setup-request`
`teardown-request`

Fields¶

Inherited:: GObject.Object (1)

Name	Type	Access	Description
parent	`GObject.Object`	r

Class Details¶

class GstRtspServer.RTSPClient(**kwargs)¶

Bases:: GObject.Object
Abstract:: No
Structure:: GstRtspServer.RTSPClientClass

The client object represents the connection and its state with a client.

classmethod new()[source]¶

Returns:: a new GstRtspServer.RTSPClient
Return type:: GstRtspServer.RTSPClient

Create a new GstRtspServer.RTSPClient instance.

attach(context)[source]¶

Parameters:: context (GLib.MainContext or None) – a GLib.MainContext
Returns:: the ID (greater than 0) for the source within the GLib.MainContext.
Return type:: int

Attaches self to context. When the mainloop for context is run, the client will be dispatched. When context is None, 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.

close()[source]¶: Close the connection of self and remove all media it was managing.

New in version 1.4.

get_auth()[source]¶

Returns:: the GstRtspServer.RTSPAuth of self. GObject.Object.unref() after usage.
Return type:: GstRtspServer.RTSPAuth or None

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

get_connection()[source]¶

Returns:: the GstRtsp.RTSPConnection of self. The connection object returned remains valid until the client is freed.
Return type:: GstRtsp.RTSPConnection or None

Get the GstRtsp.RTSPConnection of self.

get_content_length_limit()[source]¶

Returns:: the Content-Length limit.
Return type:: int

Get the Content-Length limit of self.

New in version 1.18.

get_mount_points()[source]¶

Returns:: a GstRtspServer.RTSPMountPoints, unref after usage.
Return type:: GstRtspServer.RTSPMountPoints or None

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

get_session_pool()[source]¶

Returns:: a GstRtspServer.RTSPSessionPool, unref after usage.
Return type:: GstRtspServer.RTSPSessionPool or None

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

get_stream_transport(channel)[source]¶

Parameters:: channel (int) –
Returns:: the GstRtspServer.RTSPStreamTransport associated with channel.
Return type:: GstRtspServer.RTSPStreamTransport or None

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

New in version 1.18.

get_thread_pool()[source]¶

Returns:: the GstRtspServer.RTSPThreadPool of self. GObject.Object.unref() after usage.
Return type:: GstRtspServer.RTSPThreadPool or None

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

handle_message(message)[source]¶

Parameters:: message (GstRtsp.RTSPMessage) – an GstRtsp.RTSPMessage
Returns:: a GstRtsp.RTSPResult.
Return type:: GstRtsp.RTSPResult

Let the client handle message.

send_message(session, message)[source]¶

Parameters:

session (GstRtspServer.RTSPSession or None) – a GstRtspServer.RTSPSession to send the message to or None
message (GstRtsp.RTSPMessage) – The GstRtsp.RTSPMessage to send

Return type:

GstRtsp.RTSPResult

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

session_filter(func, *user_data)[source]¶

Parameters:

func (GstRtspServer.RTSPClientSessionFilterFunc or None) – a callback
user_data (object or None) – user GstRtsp.data passed to func

Returns:

a GLib.List with all sessions for which func returned GstRtspServer.RTSPFilterResult.REF. After usage, each element in the GLib.List should be unreffed before the list is freed.

Return type:

[GstRtspServer.RTSPSession]

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

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

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

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

When func is None, GstRtspServer.RTSPFilterResult.REF will be assumed for each session.

set_auth(auth)[source]¶

Parameters:: auth (GstRtspServer.RTSPAuth or None) – a GstRtspServer.RTSPAuth

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

set_connection(conn)[source]¶

Parameters:: conn (GstRtsp.RTSPConnection) – a GstRtsp.RTSPConnection
Returns:: True on success.
Return type:: bool

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

set_content_length_limit(limit)[source]¶

Parameters:: limit (int) – Content-Length limit

Configure self to use the specified Content-Length limit.

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

New in version 1.18.

set_mount_points(mounts)[source]¶

Parameters:: mounts (GstRtspServer.RTSPMountPoints or None) – a GstRtspServer.RTSPMountPoints

Set mounts as the mount points for self 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.

set_send_func(func, *user_data)[source]¶

Parameters:

func (GstRtspServer.RTSPClientSendFunc) – a GstRtspServer.RTSPClientSendFunc
user_data (object or None) – user GstRtsp.data passed to 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.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.

set_send_messages_func(func, *user_data)[source]¶

Parameters:

func (GstRtspServer.RTSPClientSendMessagesFunc) – a GstRtspServer.RTSPClientSendMessagesFunc
user_data (object or None) – user GstRtsp.data passed to 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.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.

New in version 1.16.

set_session_pool(pool)[source]¶

Parameters:: pool (GstRtspServer.RTSPSessionPool or None) – a GstRtspServer.RTSPSessionPool

Set pool as the sessionpool for self 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.

set_thread_pool(pool)[source]¶

Parameters:: pool (GstRtspServer.RTSPThreadPool or None) – a GstRtspServer.RTSPThreadPool

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

do_adjust_error_code(ctx, code) virtual¶

Parameters:

ctx (GstRtspServer.RTSPContext) – a GstRtspServer.RTSPContext
code (GstRtsp.RTSPStatusCode) – a GstRtsp.RTSPStatusCode

Returns:

a GstRtsp.RTSPStatusCode, containing the adjusted error code.

Return type:

GstRtsp.RTSPStatusCode

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

New in version 1.22.

do_adjust_play_mode(context, range, flags, rate, trickmode_interval, enable_rate_control) virtual¶

Parameters:

context (GstRtspServer.RTSPContext) –
range (GstRtsp.RTSPTimeRange) –
flags (Gst.SeekFlags) –
rate (float) –
trickmode_interval (int) –
enable_rate_control (bool) –

Return type:

GstRtsp.RTSPStatusCode

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

New in version 1.18.

do_adjust_play_response(context) virtual¶

Parameters:: context (GstRtspServer.RTSPContext) –
Return type:: GstRtsp.RTSPStatusCode

called to give the implementation the possibility to adjust the GstRtsp.response to a play GstRtsp.request, for example if extra headers were parsed when GstRtspServer.RTSPClientClass.adjust_play_mode was called.

New in version 1.18.

do_announce_request(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –

do_check_requirements(ctx, arr) virtual¶

Parameters:

ctx (GstRtspServer.RTSPContext) –
arr (str) –

Return type:

str

do_closed() virtual¶

do_configure_client_media(media, stream, ctx) virtual¶

Parameters:

media (GstRtspServer.RTSPMedia) –
stream (GstRtspServer.RTSPStream) –
ctx (GstRtspServer.RTSPContext) –

Return type:

bool

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

do_configure_client_transport(ctx, ct) virtual¶

Parameters:

ctx (GstRtspServer.RTSPContext) –
ct (GstRtsp.RTSPTransport) –

Return type:

bool

called when the client transport needs to be configured.

do_create_sdp(media) virtual¶

Parameters:: media (GstRtspServer.RTSPMedia) –
Return type:: GstSdp.SDPMessage

called when the SDP needs to be created for media.

do_describe_request(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –

do_get_parameter_request(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –

do_handle_response(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –

do_handle_sdp(ctx, media, sdp) virtual¶

Parameters:

ctx (GstRtspServer.RTSPContext) –
media (GstRtspServer.RTSPMedia) –
sdp (GstSdp.SDPMessage) –

Return type:

bool

do_make_path_from_uri(uri) virtual¶

Parameters:: uri (GstRtsp.RTSPUrl) –
Return type:: str

called to create path from uri.

do_new_session(session) virtual¶

Parameters:: session (GstRtspServer.RTSPSession) –

do_options_request(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –

do_params_get(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –
Return type:: GstRtsp.RTSPResult

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

do_params_set(ctx) virtual¶

Parameters:: ctx (GstRtspServer.RTSPContext) –
Return type:: GstRtsp.RTSPResult

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

do_pause_request(ctx) virtual¶