srtsrc

srtsrc is a network source that reads SRT packets from the network.

Examples

 gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001" ! fakesink

This pipeline shows how to connect SRT server by setting property.

 gst-launch-1.0 -v srtsrc uri="srt://:7001?mode=listener" ! fakesink

This pipeline shows how to wait SRT connection by setting property.

 gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001?mode=rendez-vous" ! fakesink

This pipeline shows how to connect SRT server by setting property and using the rendez-vous mode.

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBaseSrc
                    ╰──GstPushSrc
                        ╰──srtsrc

Implemented interfaces

Factory details

Authors: – Justin Kim

Classification: – Source/Network

Rank – primary

Plugin – srt

Package – GStreamer Bad Plug-ins

Pad Templates

src

ANY

Presence – always

Direction – src

Object type – GstPad

Signals

caller-added 

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

A new caller has connected to srtsrc.

Parameters:

gstsrtsrc –

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

The given caller has disconnected.

Parameters:

gstsrtsrc –

the srtsrc 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

The maximum accepted transmission latency.

Flags : Read / Write

Default value : 125

localaddress 

“localaddress” gchararray

The address to bind when is listener or rendezvous. This property can be set by URI parameters.

Flags : Read / Write

Default value : NULL

localport 

“localport” guint

The local port to bind when is listener or rendezvous. This property can be set by URI parameters.

Flags : Read / Write

Default value : 7001

mode 

“mode” GstSRTConnectionMode *

The SRT connection mode. This property can be set by URI parameters.

Flags : Read / Write

Default value : caller (1)

passphrase 

“passphrase” gchararray

The password for the encrypted transmission. This property can be set by URI parameters.

Flags : Read / Write

pbkeylen 

“pbkeylen” GstSRTKeyLength *

The crypto key length. This property can be set by URI parameters.

Flags : Read / Write

Default value : no-key (0)

poll-timeout 

“poll-timeout” gint

The polling timeout used when srt poll is started. Even if the default value indicates infinite waiting, it can be cancellable according to GstState This property can be set by URI parameters.

Flags : Read / Write

Default value : -1

stats 

“stats” GstStructure *

The statistics from SRT.

Flags : Read

Default value : 

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

streamid 

“streamid” gchararray

The stream id for the SRT access control.

Flags : Read / Write

Default value : NULL

uri 

“uri” gchararray

The URI used by SRT connection. User can specify SRT specific options by URI parameters. Refer to Mediun: SRT

Flags : Read / Write

Default value : srt://127.0.0.1:7001

wait-for-connection 

“wait-for-connection” gboolean

Block the stream until a client connects

Flags : Read / Write

Default value : true

Named constants

GstSRTConnectionMode

Members

none (0) – GST_SRT_CONNECTION_MODE_NONE
caller (1) – GST_SRT_CONNECTION_MODE_CALLER
listener (2) – GST_SRT_CONNECTION_MODE_LISTENER
rendezvous (3) – GST_SRT_CONNECTION_MODE_RENDEZVOUS

GstSRTKeyLength

Members

no-key (0) – GST_SRT_KEY_LENGTH_NO_KEY
0 (0) – GST_SRT_KEY_LENGTH_0
16 (16) – GST_SRT_KEY_LENGTH_16
24 (24) – GST_SRT_KEY_LENGTH_24
32 (32) – GST_SRT_KEY_LENGTH_32

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