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 uri property.

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

This pipeline shows how to wait SRT connection by setting uri 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 uri property and using the rendez-vous mode.

Additionally, in listener mode, this element supports sharing the underlying SRT connection in order to receive multiple SRT stream using a single UDP port.

Connection sharing is acheived using the connection-key property, two receivers, in the same application, with the same connection-key value will share their SRT connection. The demultiplexing of the streams will be performed using the streamid property.

  gst-launch-1.0 -v \
    srtsrc connection-key=multiplex uri="srt://127.0.0.1:5000?mode=listener&streamid=one" ! tsparse ! queue ! filesink location=one.ts \
    srtsrc connection-key=multiplex uri="srt://127.0.0.1:5000?mode=listener&streamid=two" ! tsparse ! queue ! filesink location=two.ts
 |] This pipeline shows two SRT listeners sharing one UDP port

 |[
  gst-launch-1.0 -v mpegtsmux name=muxone alignment=7 mpegtsmux name=muxtwo alignment=7 \
    videotestsrc pattern=ball ! queue ! x264enc ! muxone. \
    videotestsrc ! queue ! x264enc ! muxtwo. \
    muxone. ! queue ! srtsink uri="srt://127.0.0.1:5000?mode=caller&streamid=one" \
    muxtwo. ! queue ! srtsink uri="srt://127.0.0.1:5000?mode=caller&streamid=two"
 |] This pipeline shows two SRT callers sending to the same port with different `streamid`

 `srtsrc` also supports listening on an endpoint without a stream ID, in which case
 the stream ID will be updated upon connection of a sender:

 ```
 gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:5000?mode=listener" ! tsparse ! queue ! filesink location=output.ts
 ```

 ```
 gst-launch-1.0 -v videotestsrc pattern=ball ! queue ! x264enc ! mpegtsmux alignment=7 ! queue ! \
   srtsink uri="srt://127.0.0.1:5000?mode=caller&streamid=one"
 ```

 You can use this in combination with connection sharing to ingest multiple streams
 on the same endpoint, by spinning up a new srtsrc in a `notify::streamid` signal handler.

 The connections RecMutex is held during emission of the signal.

Hierarchy

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

Implemented interfaces

GstURIHandler

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-connecting

gboolean
caller_connecting_callback (GstElement * gstsrtsrc,
                            GSocketAddress * addr,
                            gchararray stream_id,
                            gpointer udata)

def caller_connecting_callback (gstsrtsrc, addr, stream_id, udata):
    #python callback for the 'caller-connecting' signal

function caller_connecting_callback(gstsrtsrc: GstElement * gstsrtsrc, addr: GSocketAddress * addr, stream_id: gchararray stream_id, udata: gpointer udata): {
    // javascript callback for the 'caller-connecting' signal
}

Whether to accept or reject a caller's connection to srtsrc in listener mode. The Caller's connection is rejected if the callback returns FALSE, else the connection is accepeted.

Parameters:

gstsrtsrc –

the srtsrc element that emitted this signal

addr –

the GSocketAddress that describes the client socket

stream_id –

the stream Id to which the caller wants to connect

udata –

No description available

Returns (gboolean) –

No description available

Flags: Run Last

Since : 1.20

caller-rejected

caller_rejected_callback (GstElement * gstsrtsrc,
                          GSocketAddress * addr,
                          gchararray stream_id,
                          gpointer udata)

def caller_rejected_callback (gstsrtsrc, addr, stream_id, udata):
    #python callback for the 'caller-rejected' signal

function caller_rejected_callback(gstsrtsrc: GstElement * gstsrtsrc, addr: GSocketAddress * addr, stream_id: gchararray stream_id, udata: gpointer udata): {
    // javascript callback for the 'caller-rejected' signal
}

A caller's connection to srtsrc in listener mode has been rejected.

Parameters:

gstsrtsrc –

the srtsrc element that emitted this signal

addr –

the GSocketAddress that describes the client socket

stream_id –

the stream Id to which the caller wants to connect

udata –

No description available

Flags: Run Last

Since : 1.20

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

authentication

“authentication” gboolean

Authenticate a connection

Flags : Read / Write

Default value : false

auto-reconnect

“auto-reconnect” gboolean

Boolean to choose whether to automatically reconnect. If TRUE, an element in caller mode will try to reconnect instead of reporting an error.

This will be ignored for authentication failures.

Flags : Read / Write

Default value : true

Since : 1.22

connection-key

“connection-key” gchararray

Identifier for sharing SRT connection when listening, elements with the same connection-key will use on the same UDP socket.

Flags : Read / Write

Default value : NULL

Since : 1.30

keep-listening

“keep-listening” gboolean

If FALSE, the element will return GST_FLOW_EOS when the remote client disconnects. If TRUE, the element will keep waiting for the client to reconnect. An element message named 'connection-removed' will be sent on disconnection.

Flags : Read / Write

Default value : false

Since : 1.22

latency

“latency” gint

The maximum accepted transmission latency.

Flags : Read / Write

Default value : 125

localaddress

“localaddress” gchararray

The address to bind when mode 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 mode 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 : 1000

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

SRT connection types.

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

The results of the search are

srtsrc

Examples

Example sender- and receiver for SRT connection sharing

Hierarchy

Implemented interfaces

Factory details

Pad Templates

src

Signals

caller-added

caller-connecting

caller-rejected

caller-removed

Properties

authentication

auto-reconnect

connection-key

keep-listening

latency

localaddress

localport

mode

passphrase

pbkeylen

poll-timeout

stats

streamid

uri

wait-for-connection

Named constants

GstSRTConnectionMode

Members

GstSRTKeyLength

Members

`src`