fakeaudiosink

This element is the same as fakesink but will pretend to act as an audio sink supporting the GstStreamVolume interface. This is useful for throughput testing while creating a new pipeline or for CI purposes on machines not running a real audio daemon.

Example launch lines

 gst-launch-1.0 audiotestsrc ! fakeaudiosink

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBin
                    ╰──fakeaudiosink

Implemented interfaces

Factory details

Authors: – Philippe Normand

Classification: – Audio/Sink

Rank – none

Plugin – debugutilsbad

Package – GStreamer Bad Plug-ins

Pad Templates

sink

audio/x-raw:
         format: { F64LE, F64BE, F32LE, F32BE, S32LE, S32BE, U32LE, U32BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, S16LE, S16BE, U16LE, U16BE, S8, U8 }
           rate: [ 1, 2147483647 ]
       channels: [ 1, 2147483647 ]

Presence – always

Direction – sink

Object type – GstPad

Signals

handoff 

handoff_callback (GstElement * fakeaudiosink,
                  GstBuffer * buffer,
                  GstPad * pad,
                  gpointer udata)
def handoff_callback (fakeaudiosink, buffer, pad, udata):
    #python callback for the 'handoff' signal
function handoff_callback(fakeaudiosink: GstElement * fakeaudiosink, buffer: GstBuffer * buffer, pad: GstPad * pad, udata: gpointer udata): {
    // javascript callback for the 'handoff' signal
}

This signal gets emitted before unreffing the buffer.

Parameters:

fakeaudiosink –

the fakeaudiosink instance

buffer –

the buffer that just has been received

pad –

the pad that received it

udata –
No description available

Flags: Run Last

Since : 1.22

preroll-handoff 

preroll_handoff_callback (GstElement * fakeaudiosink,
                          GstBuffer * buffer,
                          GstPad * pad,
                          gpointer udata)
def preroll_handoff_callback (fakeaudiosink, buffer, pad, udata):
    #python callback for the 'preroll-handoff' signal
function preroll_handoff_callback(fakeaudiosink: GstElement * fakeaudiosink, buffer: GstBuffer * buffer, pad: GstPad * pad, udata: gpointer udata): {
    // javascript callback for the 'preroll-handoff' signal
}

This signal gets emitted before unreffing the buffer.

Parameters:

fakeaudiosink –

the fakeaudiosink instance

buffer –

the buffer that just has been received

pad –

the pad that received it

udata –
No description available

Flags: Run Last

Since : 1.22

Properties

async 

“async” gboolean

Go asynchronously to PAUSED

Flags : Read / Write

Default value : true

blocksize 

“blocksize” guint

Size in bytes to pull per buffer (0 = default)

Flags : Read / Write

Default value : 4096

can-activate-pull 

“can-activate-pull” gboolean

Can activate in pull mode

Flags : Read / Write

Default value : false

can-activate-push 

“can-activate-push” gboolean

Can activate in push mode

Flags : Read / Write

Default value : true

drop-out-of-segment 

“drop-out-of-segment” gboolean

Drop and don't render / hand off out-of-segment buffers

Flags : Read / Write

Default value : true

dump 

“dump” gboolean

Dump buffer contents to stdout

Flags : Read / Write

Default value : false

enable-last-sample 

“enable-last-sample” gboolean

Enable the last-sample property

Flags : Read / Write

Default value : true

last-message 

“last-message” gchararray

The message describing current status

Flags : Read

Default value : NULL

last-sample 

“last-sample” GstSample *

The last sample received in the sink

Flags : Read

max-bitrate 

“max-bitrate” guint64

The maximum bits per second to render (0 = disabled)

Flags : Read / Write

Default value : 0

max-lateness 

“max-lateness” gint64

Maximum number of nanoseconds that a buffer can be late before it is dropped (-1 unlimited)

Flags : Read / Write

Default value : 18446744073709551615

mute 

“mute” gboolean

Control the mute state

Flags : Read / Write

Default value : false

Since : 1.20

num-buffers 

“num-buffers” gint

Number of buffers to accept going EOS

Flags : Read / Write

Default value : -1

processing-deadline 

“processing-deadline” guint64

Maximum processing time for a buffer in nanoseconds

Flags : Read / Write

Default value : 20000000

qos 

“qos” gboolean

Generate Quality-of-Service events upstream

Flags : Read / Write

Default value : true

render-delay 

“render-delay” guint64

Additional render delay of the sink in nanoseconds

Flags : Read / Write

Default value : 0

signal-handoffs 

“signal-handoffs” gboolean

Send a signal before unreffing the buffer

Flags : Read / Write

Default value : false

silent 

“silent” gboolean

Don't produce last_message events

Flags : Read / Write

Default value : true

state-error 

“state-error” Fake-audio-sink-state-error *

Generate a state change error

Flags : Read / Write

Default value : none (0)

stats 

“stats” GstStructure *

Sink Statistics

Flags : Read

Default value : 

application/x-gst-base-sink-stats, average-rate=(double)0, dropped=(guint64)0, rendered=(guint64)0;

sync 

“sync” gboolean

Sync on the clock

Flags : Read / Write

Default value : true

throttle-time 

“throttle-time” guint64

The time to keep between rendered buffers (0 = disabled)

Flags : Read / Write

Default value : 0

ts-offset 

“ts-offset” gint64

Timestamp offset in nanoseconds

Flags : Read / Write

Default value : 0

volume 

“volume” gdouble

Control the audio volume

Flags : Read / Write

Default value : 1

Since : 1.20

Named constants

Fake-audio-sink-state-error

Proxy for GstFakeSinkError.

Members

none (0) – No state change errors
null-to-ready (1) – Fail state change from NULL to READY
ready-to-paused (2) – Fail state change from READY to PAUSED
paused-to-playing (3) – Fail state change from PAUSED to PLAYING
playing-to-paused (4) – Fail state change from PLAYING to PAUSED
paused-to-ready (5) – Fail state change from PAUSED to READY
ready-to-null (6) – Fail state change from READY to NULL

Since : 1.22

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