srtsink

srtsink is a network sink that sends SRT packets to the network.

Examples

 gst-launch-1.0 -v audiotestsrc ! srtsink uri=srt://host

This pipeline shows how to serve SRT packets through the default port.

 gst-launch-1.0 -v audiotestsrc ! srtsink uri=srt://:port

This pipeline shows how to wait SRT callers.

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBaseSink
                    ╰──srtsink

Implemented interfaces

Factory details

Authors: – Justin Kim

Classification: – Sink/Network

Rank – primary

Plugin – srt

Package – GStreamer Bad Plug-ins

Pad Templates

sink

ANY

Presence – always

Direction – sink

Object type – GstPad

Signals

caller-added 

caller_added_callback (GstElement * gstsrtsink,
                       gint unused,
                       GSocketAddress * addr,
                       gpointer udata)
def caller_added_callback (gstsrtsink, unused, addr, udata):
    #python callback for the 'caller-added' signal
function caller_added_callback(gstsrtsink: GstElement * gstsrtsink, unused: gint unused, addr: GSocketAddress * addr, udata: gpointer udata): {
    // javascript callback for the 'caller-added' signal
}

A new caller has connected to gstsrtsink.

Parameters:

gstsrtsink –

the srtsink element that emitted this signal

unused –

always zero (for ABI compatibility with previous versions)

addr –

the GSocketAddress of the new caller

udata –
No description available

Flags: Run Last

caller-removed 

caller_removed_callback (GstElement * gstsrtsink,
                         gint unused,
                         GSocketAddress * addr,
                         gpointer udata)
def caller_removed_callback (gstsrtsink, unused, addr, udata):
    #python callback for the 'caller-removed' signal
function caller_removed_callback(gstsrtsink: GstElement * gstsrtsink, unused: gint unused, addr: GSocketAddress * addr, udata: gpointer udata): {
    // javascript callback for the 'caller-removed' signal
}

The given caller has disconnected.

Parameters:

gstsrtsink –

the srtsink element that emitted this signal

unused –

always zero (for ABI compatibility with previous versions)

addr –

the GSocketAddress of the caller

udata –
No description available

Flags: Run Last

Properties

latency 

“latency” gint

Minimum latency (milliseconds)

Flags : Read / Write

Default value : 125

localaddress 

“localaddress” gchararray

Local address to bind

Flags : Read / Write

Default value : NULL

localport 

“localport” guint

Local port to bind

Flags : Read / Write

Default value : 7001

mode 

“mode” GstSRTConnectionMode *

SRT connection mode

Flags : Read / Write

Default value : caller (1)

passphrase 

“passphrase” gchararray

Password for the encrypted transmission

Flags : Read / Write

pbkeylen 

“pbkeylen” GstSRTKeyLength *

Crypto key length in bytes

Flags : Read / Write

Default value : no-key (0)

poll-timeout 

“poll-timeout” gint

Return poll wait after timeout milliseconds (-1 = infinite)

Flags : Read / Write

Default value : 1000

stats 

“stats” GstStructure *

SRT Statistics

Flags : Read

Default value : 

application/x-srt-statistics, bytes-sent-total=(guint64)0;

streamid 

“streamid” gchararray

Stream ID for the SRT access control

Flags : Read / Write

Default value : NULL

uri 

“uri” gchararray

URI in the form of srt://address:port

Flags : Read / Write

Default value : srt://127.0.0.1:7001

wait-for-connection 

“wait-for-connection” gboolean

Boolean to block streaming until a client connects. If TRUE, `srtsink' will stream only when a client is connected.

Flags : Read / Write

Default value : true

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