rtsp stream transport

The GstRTSPStreamTransport configures the transport used by a GstRTSPStream. It is usually manages by a GstRTSPSessionMedia object.

With gst_rtsp_stream_transport_set_callbacks, callbacks can be configured to handle the RTP and RTCP packets from the stream, for example when they need to be sent over TCP.

With gst_rtsp_stream_transport_set_active the transports are added and removed from the stream.

A GstRTSPStream will call gst_rtsp_stream_transport_keep_alive when RTCP is received from the client. It will also call gst_rtsp_stream_transport_set_timed_out when a receiver has timed out.

A GstRTSPClient will call gst_rtsp_stream_transport_message_sent when it has sent a data message for the transport.

Last reviewed on 2013-07-16 (1.0.0)

GstRTSPStreamTransport

GObject
    ╰──GstRTSPStreamTransport

A Transport description for a stream

Members

parent (GObject) –

parent instance


Class structure

GstRTSPStreamTransportClass

Fields
parent_class (GObjectClass) –
No description available

GstRtspServer.RTSPStreamTransportClass

Attributes
parent_class (GObject.ObjectClass) –
No description available

GstRtspServer.RTSPStreamTransportClass

Attributes
parent_class (GObject.ObjectClass) –
No description available

GstRtspServer.RTSPStreamTransport

GObject.Object
    ╰──GstRtspServer.RTSPStreamTransport

A Transport description for a stream

Members

parent (GObject.Object) –

parent instance


GstRtspServer.RTSPStreamTransport

GObject.Object
    ╰──GstRtspServer.RTSPStreamTransport

A Transport description for a stream

Members

parent (GObject.Object) –

parent instance


Constructors

gst_rtsp_stream_transport_new

GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream,
                               GstRTSPTransport * tr)

Create a new GstRTSPStreamTransport that can be used to manage stream with transport tr.

Parameters:

stream

a GstRTSPStream

tr ( [transfer: full])

a GstRTSPTransport

Returns ( [transfer: full])

a new GstRTSPStreamTransport


GstRtspServer.RTSPStreamTransport.prototype.new

function GstRtspServer.RTSPStreamTransport.prototype.new(stream: GstRtspServer.RTSPStream, tr: GstRtsp.RTSPTransport): {
    // javascript wrapper for 'gst_rtsp_stream_transport_new'
}

Create a new GstRtspServer.RTSPStreamTransport that can be used to manage stream with transport tr.

Parameters:

a GstRTSPTransport


GstRtspServer.RTSPStreamTransport.new

def GstRtspServer.RTSPStreamTransport.new (stream, tr):
    #python wrapper for 'gst_rtsp_stream_transport_new'

Create a new GstRtspServer.RTSPStreamTransport that can be used to manage stream with transport tr.

Parameters:

a GstRTSPTransport


Methods

gst_rtsp_stream_transport_get_rtpinfo

gchar *
gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
                                       GstClockTime start_time)

Get the RTP-Info string for trans and start_time.

Parameters:

start_time

a star time

Returns ( [transfer: full][nullable])

the RTPInfo string for trans and start_time or NULL when the RTP-Info could not be determined. g_free after usage.


GstRtspServer.RTSPStreamTransport.prototype.get_rtpinfo

function GstRtspServer.RTSPStreamTransport.prototype.get_rtpinfo(start_time: Number): {
    // javascript wrapper for 'gst_rtsp_stream_transport_get_rtpinfo'
}

Get the RTP-Info string for trans and start_time.

Parameters:

start_time (Number)

a star time

Returns (String)

the RTPInfo string for trans and start_time or null when the RTP-Info could not be determined. GLib.prototype.free after usage.


GstRtspServer.RTSPStreamTransport.get_rtpinfo

def GstRtspServer.RTSPStreamTransport.get_rtpinfo (self, start_time):
    #python wrapper for 'gst_rtsp_stream_transport_get_rtpinfo'

Get the RTP-Info string for trans and start_time.

Parameters:

start_time (int)

a star time

Returns (str)

the RTPInfo string for trans and start_time or None when the RTP-Info could not be determined. GLib.free after usage.


gst_rtsp_stream_transport_get_stream

GstRTSPStream *
gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)

Get the GstRTSPStream used when constructing trans.

Parameters:

Returns ( [transfer: none][nullable])

the stream used when constructing trans.


GstRtspServer.RTSPStreamTransport.prototype.get_stream

function GstRtspServer.RTSPStreamTransport.prototype.get_stream(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_get_stream'
}

Get the GstRtspServer.RTSPStream used when constructing trans.

Returns (GstRtspServer.RTSPStream)

the stream used when constructing trans.


GstRtspServer.RTSPStreamTransport.get_stream

def GstRtspServer.RTSPStreamTransport.get_stream (self):
    #python wrapper for 'gst_rtsp_stream_transport_get_stream'

Get the GstRtspServer.RTSPStream used when constructing trans.

Returns (GstRtspServer.RTSPStream)

the stream used when constructing trans.


gst_rtsp_stream_transport_get_transport

const GstRTSPTransport *
gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)

Get the transport configured in trans.

Parameters:

Returns ( [transfer: none][nullable])

the transport configured in trans. It remains valid for as long as trans is valid.


GstRtspServer.RTSPStreamTransport.prototype.get_transport

function GstRtspServer.RTSPStreamTransport.prototype.get_transport(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_get_transport'
}

Get the transport configured in trans.

Returns (GstRtsp.RTSPTransport)

the transport configured in trans. It remains valid for as long as trans is valid.


GstRtspServer.RTSPStreamTransport.get_transport

def GstRtspServer.RTSPStreamTransport.get_transport (self):
    #python wrapper for 'gst_rtsp_stream_transport_get_transport'

Get the transport configured in trans.

Returns (GstRtsp.RTSPTransport)

the transport configured in trans. It remains valid for as long as trans is valid.


gst_rtsp_stream_transport_get_url

const GstRTSPUrl *
gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)

Get the url configured in trans.

Parameters:

Returns ( [transfer: none][nullable])

the url configured in trans. It remains valid for as long as trans is valid.


GstRtspServer.RTSPStreamTransport.prototype.get_url

function GstRtspServer.RTSPStreamTransport.prototype.get_url(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_get_url'
}

Get the url configured in trans.

Returns (GstRtsp.RTSPUrl)

the url configured in trans. It remains valid for as long as trans is valid.


GstRtspServer.RTSPStreamTransport.get_url

def GstRtspServer.RTSPStreamTransport.get_url (self):
    #python wrapper for 'gst_rtsp_stream_transport_get_url'

Get the url configured in trans.

Returns (GstRtsp.RTSPUrl)

the url configured in trans. It remains valid for as long as trans is valid.


gst_rtsp_stream_transport_is_timed_out

gboolean
gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)

Check if trans is timed out.

Parameters:

Returns

TRUE if trans timed out.


GstRtspServer.RTSPStreamTransport.prototype.is_timed_out

function GstRtspServer.RTSPStreamTransport.prototype.is_timed_out(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_is_timed_out'
}

Check if trans is timed out.

Returns (Number)

true if trans timed out.


GstRtspServer.RTSPStreamTransport.is_timed_out

def GstRtspServer.RTSPStreamTransport.is_timed_out (self):
    #python wrapper for 'gst_rtsp_stream_transport_is_timed_out'

Check if trans is timed out.

Returns (bool)

True if trans timed out.


gst_rtsp_stream_transport_keep_alive

gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)

Signal the installed keep_alive callback for trans.

Parameters:


GstRtspServer.RTSPStreamTransport.prototype.keep_alive

function GstRtspServer.RTSPStreamTransport.prototype.keep_alive(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_keep_alive'
}

Signal the installed keep_alive callback for trans.


GstRtspServer.RTSPStreamTransport.keep_alive

def GstRtspServer.RTSPStreamTransport.keep_alive (self):
    #python wrapper for 'gst_rtsp_stream_transport_keep_alive'

Signal the installed keep_alive callback for trans.


gst_rtsp_stream_transport_message_sent

gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport * trans)

Signal the installed message_sent / message_sent_full callback for trans.

Parameters:

Since : 1.16


GstRtspServer.RTSPStreamTransport.prototype.message_sent

function GstRtspServer.RTSPStreamTransport.prototype.message_sent(): {
    // javascript wrapper for 'gst_rtsp_stream_transport_message_sent'
}

Signal the installed message_sent / message_sent_full callback for trans.

Since : 1.16


GstRtspServer.RTSPStreamTransport.message_sent

def GstRtspServer.RTSPStreamTransport.message_sent (self):
    #python wrapper for 'gst_rtsp_stream_transport_message_sent'

Signal the installed message_sent / message_sent_full callback for trans.

Since : 1.16


gst_rtsp_stream_transport_recv_data

GstFlowReturn
gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport * trans,
                                     guint channel,
                                     GstBuffer * buffer)

Receive buffer on channel trans.

Parameters:

channel

a channel

buffer ( [transfer: full])

a GstBuffer

Returns

a GstFlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.


GstRtspServer.RTSPStreamTransport.prototype.recv_data

function GstRtspServer.RTSPStreamTransport.prototype.recv_data(channel: Number, buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtsp_stream_transport_recv_data'
}

Receive buffer on channel trans.

Parameters:

channel (Number)

a channel

buffer (Gst.Buffer)

a Gst.Buffer

Returns (Gst.FlowReturn)

a Gst.FlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.


GstRtspServer.RTSPStreamTransport.recv_data

def GstRtspServer.RTSPStreamTransport.recv_data (self, channel, buffer):
    #python wrapper for 'gst_rtsp_stream_transport_recv_data'

Receive buffer on channel trans.

Parameters:

channel (int)

a channel

buffer (Gst.Buffer)

a Gst.Buffer

Returns (Gst.FlowReturn)

a Gst.FlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.


gst_rtsp_stream_transport_send_rtcp

gboolean
gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
                                     GstBuffer * buffer)

Send buffer to the installed RTCP callback for trans.

Parameters:

buffer ( [transfer: none])

a GstBuffer

Returns

TRUE on success


GstRtspServer.RTSPStreamTransport.prototype.send_rtcp

function GstRtspServer.RTSPStreamTransport.prototype.send_rtcp(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtsp_stream_transport_send_rtcp'
}

Send buffer to the installed RTCP callback for trans.

Returns (Number)

true on success


GstRtspServer.RTSPStreamTransport.send_rtcp

def GstRtspServer.RTSPStreamTransport.send_rtcp (self, buffer):
    #python wrapper for 'gst_rtsp_stream_transport_send_rtcp'

Send buffer to the installed RTCP callback for trans.

Returns (bool)

True on success


gst_rtsp_stream_transport_send_rtcp_list

gboolean
gst_rtsp_stream_transport_send_rtcp_list (GstRTSPStreamTransport * trans,
                                          GstBufferList * buffer_list)

Send buffer_list to the installed RTCP callback for trans.

Parameters:

buffer_list ( [transfer: none])

a GstBuffer

Returns

TRUE on success

Since : 1.16


GstRtspServer.RTSPStreamTransport.prototype.send_rtcp_list

function GstRtspServer.RTSPStreamTransport.prototype.send_rtcp_list(buffer_list: Gst.BufferList): {
    // javascript wrapper for 'gst_rtsp_stream_transport_send_rtcp_list'
}

Send buffer_list to the installed RTCP callback for trans.

Returns (Number)

true on success

Since : 1.16


GstRtspServer.RTSPStreamTransport.send_rtcp_list

def GstRtspServer.RTSPStreamTransport.send_rtcp_list (self, buffer_list):
    #python wrapper for 'gst_rtsp_stream_transport_send_rtcp_list'

Send buffer_list to the installed RTCP callback for trans.

Returns (bool)

True on success

Since : 1.16


gst_rtsp_stream_transport_send_rtp

gboolean
gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
                                    GstBuffer * buffer)

Send buffer to the installed RTP callback for trans.

Parameters:

buffer ( [transfer: none])

a GstBuffer

Returns

TRUE on success


GstRtspServer.RTSPStreamTransport.prototype.send_rtp

function GstRtspServer.RTSPStreamTransport.prototype.send_rtp(buffer: Gst.Buffer): {
    // javascript wrapper for 'gst_rtsp_stream_transport_send_rtp'
}

Send buffer to the installed RTP callback for trans.

Returns (Number)

true on success


GstRtspServer.RTSPStreamTransport.send_rtp

def GstRtspServer.RTSPStreamTransport.send_rtp (self, buffer):
    #python wrapper for 'gst_rtsp_stream_transport_send_rtp'

Send buffer to the installed RTP callback for trans.

Returns (bool)

True on success


gst_rtsp_stream_transport_send_rtp_list

gboolean
gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport * trans,
                                         GstBufferList * buffer_list)

Send buffer_list to the installed RTP callback for trans.

Parameters:

buffer_list ( [transfer: none])

a GstBufferList

Returns

TRUE on success

Since : 1.16


GstRtspServer.RTSPStreamTransport.prototype.send_rtp_list

function GstRtspServer.RTSPStreamTransport.prototype.send_rtp_list(buffer_list: Gst.BufferList): {
    // javascript wrapper for 'gst_rtsp_stream_transport_send_rtp_list'
}

Send buffer_list to the installed RTP callback for trans.

Returns (Number)

true on success

Since : 1.16


GstRtspServer.RTSPStreamTransport.send_rtp_list

def GstRtspServer.RTSPStreamTransport.send_rtp_list (self, buffer_list):
    #python wrapper for 'gst_rtsp_stream_transport_send_rtp_list'

Send buffer_list to the installed RTP callback for trans.

Returns (bool)

True on success

Since : 1.16


gst_rtsp_stream_transport_set_active

gboolean
gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
                                      gboolean active)

Activate or deactivate datatransfer configured in trans.

Parameters:

active

new state of trans

Returns

TRUE when the state was changed.


GstRtspServer.RTSPStreamTransport.prototype.set_active

function GstRtspServer.RTSPStreamTransport.prototype.set_active(active: Number): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_active'
}

Activate or deactivate datatransfer configured in trans.

Parameters:

active (Number)

new state of trans

Returns (Number)

true when the state was changed.


GstRtspServer.RTSPStreamTransport.set_active

def GstRtspServer.RTSPStreamTransport.set_active (self, active):
    #python wrapper for 'gst_rtsp_stream_transport_set_active'

Activate or deactivate datatransfer configured in trans.

Parameters:

active (bool)

new state of trans

Returns (bool)

True when the state was changed.


gst_rtsp_stream_transport_set_callbacks

gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
                                         GstRTSPSendFunc send_rtp,
                                         GstRTSPSendFunc send_rtcp,
                                         gpointer user_data,
                                         GDestroyNotify notify)

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

send_rtp ( [scope notified])

a callback called when RTP should be sent

send_rtcp ( [scope notified])

a callback called when RTCP should be sent

user_data ( [closure])

user data passed to callbacks

notify ( [allow-none])

called with the user_data when no longer needed.


GstRtspServer.RTSPStreamTransport.prototype.set_callbacks

function GstRtspServer.RTSPStreamTransport.prototype.set_callbacks(send_rtp: GstRtspServer.RTSPSendFunc, send_rtcp: GstRtspServer.RTSPSendFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_callbacks'
}

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

a callback called when RTP should be sent

send_rtcp (GstRtspServer.RTSPSendFunc)

a callback called when RTCP should be sent

user_data (Object)

user data passed to callbacks


GstRtspServer.RTSPStreamTransport.set_callbacks

def GstRtspServer.RTSPStreamTransport.set_callbacks (self, send_rtp, send_rtcp, *user_data):
    #python wrapper for 'gst_rtsp_stream_transport_set_callbacks'

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

a callback called when RTP should be sent

send_rtcp (GstRtspServer.RTSPSendFunc)

a callback called when RTCP should be sent

user_data (variadic)

user data passed to callbacks


gst_rtsp_stream_transport_set_keepalive

gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
                                         GstRTSPKeepAliveFunc keep_alive,
                                         gpointer user_data,
                                         GDestroyNotify notify)

Install callbacks that will be called when RTCP packets are received from the receiver of trans.

Parameters:

keep_alive ( [scope notified])

a callback called when the receiver is active

user_data ( [closure])

user data passed to callback

notify ( [allow-none])

called with the user_data when no longer needed.


GstRtspServer.RTSPStreamTransport.prototype.set_keepalive

function GstRtspServer.RTSPStreamTransport.prototype.set_keepalive(keep_alive: GstRtspServer.RTSPKeepAliveFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_keepalive'
}

Install callbacks that will be called when RTCP packets are received from the receiver of trans.

Parameters:

a callback called when the receiver is active

user_data (Object)

user data passed to callback


GstRtspServer.RTSPStreamTransport.set_keepalive

def GstRtspServer.RTSPStreamTransport.set_keepalive (self, keep_alive, *user_data):
    #python wrapper for 'gst_rtsp_stream_transport_set_keepalive'

Install callbacks that will be called when RTCP packets are received from the receiver of trans.

Parameters:

a callback called when the receiver is active

user_data (variadic)

user data passed to callback


gst_rtsp_stream_transport_set_list_callbacks

gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport * trans,
                                              GstRTSPSendListFunc send_rtp_list,
                                              GstRTSPSendListFunc send_rtcp_list,
                                              gpointer user_data,
                                              GDestroyNotify notify)

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

send_rtp_list ( [scope notified])

a callback called when RTP should be sent

send_rtcp_list ( [scope notified])

a callback called when RTCP should be sent

user_data ( [closure])

user data passed to callbacks

notify ( [allow-none])

called with the user_data when no longer needed.

Since : 1.16


GstRtspServer.RTSPStreamTransport.prototype.set_list_callbacks

function GstRtspServer.RTSPStreamTransport.prototype.set_list_callbacks(send_rtp_list: GstRtspServer.RTSPSendListFunc, send_rtcp_list: GstRtspServer.RTSPSendListFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_list_callbacks'
}

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

send_rtp_list (GstRtspServer.RTSPSendListFunc)

a callback called when RTP should be sent

send_rtcp_list (GstRtspServer.RTSPSendListFunc)

a callback called when RTCP should be sent

user_data (Object)

user data passed to callbacks

Since : 1.16


GstRtspServer.RTSPStreamTransport.set_list_callbacks

def GstRtspServer.RTSPStreamTransport.set_list_callbacks (self, send_rtp_list, send_rtcp_list, *user_data):
    #python wrapper for 'gst_rtsp_stream_transport_set_list_callbacks'

Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.

Parameters:

send_rtp_list (GstRtspServer.RTSPSendListFunc)

a callback called when RTP should be sent

send_rtcp_list (GstRtspServer.RTSPSendListFunc)

a callback called when RTCP should be sent

user_data (variadic)

user data passed to callbacks

Since : 1.16


gst_rtsp_stream_transport_set_message_sent

gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport * trans,
                                            GstRTSPMessageSentFunc message_sent,
                                            gpointer user_data,
                                            GDestroyNotify notify)

Install a callback that will be called when a message has been sent on trans.

Parameters:

message_sent ( [scope notified])

a callback called when a message has been sent

user_data ( [closure])

user data passed to callback

notify ( [allow-none])

called with the user_data when no longer needed


GstRtspServer.RTSPStreamTransport.prototype.set_message_sent

function GstRtspServer.RTSPStreamTransport.prototype.set_message_sent(message_sent: GstRtspServer.RTSPMessageSentFunc, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_message_sent'
}

Install a callback that will be called when a message has been sent on trans.

Parameters:

a callback called when a message has been sent

user_data (Object)

user data passed to callback


GstRtspServer.RTSPStreamTransport.set_message_sent

def GstRtspServer.RTSPStreamTransport.set_message_sent (self, message_sent, *user_data):
    #python wrapper for 'gst_rtsp_stream_transport_set_message_sent'

Install a callback that will be called when a message has been sent on trans.

Parameters:

a callback called when a message has been sent

user_data (variadic)

user data passed to callback


gst_rtsp_stream_transport_set_message_sent_full

gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport * trans,
                                                 GstRTSPMessageSentFuncFull message_sent,
                                                 gpointer user_data,
                                                 GDestroyNotify notify)

Install a callback that will be called when a message has been sent on trans.

Parameters:

message_sent ( [scope notified])

a callback called when a message has been sent

user_data ( [closure])

user data passed to callback

notify ( [allow-none])

called with the user_data when no longer needed

Since : 1.18


GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full

function GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full(message_sent: GstRtspServer.RTSPMessageSentFuncFull, user_data: Object): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_message_sent_full'
}

Install a callback that will be called when a message has been sent on trans.

Parameters:

a callback called when a message has been sent

user_data (Object)

user data passed to callback

Since : 1.18


GstRtspServer.RTSPStreamTransport.set_message_sent_full

def GstRtspServer.RTSPStreamTransport.set_message_sent_full (self, message_sent, *user_data):
    #python wrapper for 'gst_rtsp_stream_transport_set_message_sent_full'

Install a callback that will be called when a message has been sent on trans.

Parameters:

a callback called when a message has been sent

user_data (variadic)

user data passed to callback

Since : 1.18


gst_rtsp_stream_transport_set_timed_out

gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
                                         gboolean timedout)

Set the timed out state of trans to timedout

Parameters:

timedout

timed out value


GstRtspServer.RTSPStreamTransport.prototype.set_timed_out

function GstRtspServer.RTSPStreamTransport.prototype.set_timed_out(timedout: Number): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_timed_out'
}

Set the timed out state of trans to timedout

Parameters:

timedout (Number)

timed out value


GstRtspServer.RTSPStreamTransport.set_timed_out

def GstRtspServer.RTSPStreamTransport.set_timed_out (self, timedout):
    #python wrapper for 'gst_rtsp_stream_transport_set_timed_out'

Set the timed out state of trans to timedout

Parameters:

timedout (bool)

timed out value


gst_rtsp_stream_transport_set_transport

gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
                                         GstRTSPTransport * tr)

Set tr as the client transport. This function takes ownership of the passed tr.

Parameters:

tr ( [transfer: full])

a client GstRTSPTransport


GstRtspServer.RTSPStreamTransport.prototype.set_transport

function GstRtspServer.RTSPStreamTransport.prototype.set_transport(tr: GstRtsp.RTSPTransport): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_transport'
}

Set tr as the client transport. This function takes ownership of the passed tr.


GstRtspServer.RTSPStreamTransport.set_transport

def GstRtspServer.RTSPStreamTransport.set_transport (self, tr):
    #python wrapper for 'gst_rtsp_stream_transport_set_transport'

Set tr as the client transport. This function takes ownership of the passed tr.


gst_rtsp_stream_transport_set_url

gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
                                   const GstRTSPUrl * url)

Set url as the client url.

Parameters:

url ( [transfer: none][nullable])

a client GstRTSPUrl


GstRtspServer.RTSPStreamTransport.prototype.set_url

function GstRtspServer.RTSPStreamTransport.prototype.set_url(url: GstRtsp.RTSPUrl): {
    // javascript wrapper for 'gst_rtsp_stream_transport_set_url'
}

Set url as the client url.


GstRtspServer.RTSPStreamTransport.set_url

def GstRtspServer.RTSPStreamTransport.set_url (self, url):
    #python wrapper for 'gst_rtsp_stream_transport_set_url'

Set url as the client url.


Function Macros

GST_RTSP_STREAM_TRANSPORT_CAST

#define GST_RTSP_STREAM_TRANSPORT_CAST(obj)         ((GstRTSPStreamTransport*)(obj))

GST_RTSP_STREAM_TRANSPORT_CLASS_CAST

#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))

Callbacks

GstRTSPKeepAliveFunc

(*GstRTSPKeepAliveFunc) (gpointer user_data)

Function registered with gst_rtsp_stream_transport_set_keepalive and called when the stream is active.

Parameters:

user_data

user data


GstRtspServer.RTSPKeepAliveFunc

function GstRtspServer.RTSPKeepAliveFunc(user_data: Object): {
    // javascript wrapper for 'GstRTSPKeepAliveFunc'
}

Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_keepalive and called when the stream is active.

Parameters:

user_data (Object)

user data


GstRtspServer.RTSPKeepAliveFunc

def GstRtspServer.RTSPKeepAliveFunc (*user_data):
    #python wrapper for 'GstRTSPKeepAliveFunc'

Function registered with GstRtspServer.RTSPStreamTransport.set_keepalive and called when the stream is active.

Parameters:

user_data (variadic)

user data


GstRTSPMessageSentFunc

(*GstRTSPMessageSentFunc) (gpointer user_data)

Function registered with gst_rtsp_stream_transport_set_message_sent and called when a message has been sent on the transport.

Parameters:

user_data

user data


GstRtspServer.RTSPMessageSentFunc

function GstRtspServer.RTSPMessageSentFunc(user_data: Object): {
    // javascript wrapper for 'GstRTSPMessageSentFunc'
}

Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_message_sent and called when a message has been sent on the transport.

Parameters:

user_data (Object)

user data


GstRtspServer.RTSPMessageSentFunc

def GstRtspServer.RTSPMessageSentFunc (*user_data):
    #python wrapper for 'GstRTSPMessageSentFunc'

Function registered with GstRtspServer.RTSPStreamTransport.set_message_sent and called when a message has been sent on the transport.

Parameters:

user_data (variadic)

user data


GstRTSPMessageSentFuncFull

(*GstRTSPMessageSentFuncFull) (GstRTSPStreamTransport * trans,
                               gpointer user_data)

Function registered with gst_rtsp_stream_transport_set_message_sent_full and called when a message has been sent on the transport.

Parameters:

trans
No description available
user_data

user data

Since : 1.18


GstRtspServer.RTSPMessageSentFuncFull

function GstRtspServer.RTSPMessageSentFuncFull(trans: GstRtspServer.RTSPStreamTransport, user_data: Object): {
    // javascript wrapper for 'GstRTSPMessageSentFuncFull'
}

Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full and called when a message has been sent on the transport.

Parameters:

No description available
user_data (Object)

user data

Since : 1.18


GstRtspServer.RTSPMessageSentFuncFull

def GstRtspServer.RTSPMessageSentFuncFull (trans, *user_data):
    #python wrapper for 'GstRTSPMessageSentFuncFull'

Function registered with GstRtspServer.RTSPStreamTransport.set_message_sent_full and called when a message has been sent on the transport.

Parameters:

No description available
user_data (variadic)

user data

Since : 1.18


GstRTSPSendFunc

gboolean
(*GstRTSPSendFunc) (GstBuffer * buffer,
                    guint8 channel,
                    gpointer user_data)

Function registered with gst_rtsp_stream_transport_set_callbacks and called when buffer must be sent on channel.

Parameters:

buffer

a GstBuffer

channel

a channel

user_data

user data

Returns

TRUE on success


GstRtspServer.RTSPSendFunc

function GstRtspServer.RTSPSendFunc(buffer: Gst.Buffer, channel: Number, user_data: Object): {
    // javascript wrapper for 'GstRTSPSendFunc'
}

Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_callbacks and called when buffer must be sent on channel.

Parameters:

buffer (Gst.Buffer)

a Gst.Buffer

channel (Number)

a channel

user_data (Object)

user data

Returns (Number)

true on success


GstRtspServer.RTSPSendFunc

def GstRtspServer.RTSPSendFunc (buffer, channel, *user_data):
    #python wrapper for 'GstRTSPSendFunc'

Function registered with GstRtspServer.RTSPStreamTransport.set_callbacks and called when buffer must be sent on channel.

Parameters:

buffer (Gst.Buffer)

a Gst.Buffer

channel (int)

a channel

user_data (variadic)

user data

Returns (bool)

True on success


GstRTSPSendListFunc

gboolean
(*GstRTSPSendListFunc) (GstBufferList * buffer_list,
                        guint8 channel,
                        gpointer user_data)

Function registered with gst_rtsp_stream_transport_set_callbacks and called when buffer_list must be sent on channel.

Parameters:

buffer_list

a GstBufferList

channel

a channel

user_data

user data

Returns

TRUE on success

Since : 1.16


GstRtspServer.RTSPSendListFunc

function GstRtspServer.RTSPSendListFunc(buffer_list: Gst.BufferList, channel: Number, user_data: Object): {
    // javascript wrapper for 'GstRTSPSendListFunc'
}

Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_callbacks and called when buffer_list must be sent on channel.

Parameters:

buffer_list (Gst.BufferList)

a Gst.BufferList

channel (Number)

a channel

user_data (Object)

user data

Returns (Number)

true on success

Since : 1.16


GstRtspServer.RTSPSendListFunc

def GstRtspServer.RTSPSendListFunc (buffer_list, channel, *user_data):
    #python wrapper for 'GstRTSPSendListFunc'

Function registered with GstRtspServer.RTSPStreamTransport.set_callbacks and called when buffer_list must be sent on channel.

Parameters:

buffer_list (Gst.BufferList)

a Gst.BufferList

channel (int)

a channel

user_data (variadic)

user data

Returns (bool)

True on success

Since : 1.16


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