# GstAudioBaseSink

This is the base class for audio sinks. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of writing samples to the ringbuffer, synchronisation, clipping and flushing.

## GstAudioBaseSink

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstElement
                    ╰──GstBaseSink
                        ╰──GstAudioBaseSink
                            ╰──GstAudioSink

Opaque GstAudioBaseSink.

### Members

`element` (GstBaseSink) –
No description available
`ringbuffer` (GstAudioRingBuffer *) –
No description available
`buffer_time` (guint64) –
No description available
`latency_time` (guint64) –
No description available
`next_sample` (guint64) –
No description available
`provided_clock` (GstClock *) –
No description available
`eos_rendering` (gboolean) –
No description available

### Class structure

#### GstAudioBaseSinkClass

GstAudioBaseSink class. Override the vmethod to implement functionality.

##### Fields

`parent_class` (GstBaseSinkClass) –

the parent class.

### GstAudio.AudioBaseSinkClass

GstAudio.AudioBaseSink class. Override the vmethod to implement functionality.

##### Attributes

`parent_class` (GstBase.BaseSinkClass) –

the parent class.

### GstAudio.AudioBaseSinkClass

GstAudio.AudioBaseSink class. Override the vmethod to implement functionality.

##### Attributes

`parent_class` (GstBase.BaseSinkClass) –

the parent class.

## GstAudio.AudioBaseSink

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──Gst.Element
                    ╰──GstBase.BaseSink
                        ╰──GstAudio.AudioBaseSink
                            ╰──GstAudio.AudioSink

Opaque GstAudio.AudioBaseSink.

### Members

`element` (GstBase.BaseSink) –
No description available
`ringbuffer` (GstAudio.AudioRingBuffer) –
No description available
`buffer_time` (Number) –
No description available
`latency_time` (Number) –
No description available
`next_sample` (Number) –
No description available
`provided_clock` (Gst.Clock) –
No description available
`eos_rendering` (Number) –
No description available

## GstAudio.AudioBaseSink

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──Gst.Element
                    ╰──GstBase.BaseSink
                        ╰──GstAudio.AudioBaseSink
                            ╰──GstAudio.AudioSink

Opaque GstAudio.AudioBaseSink.

### Members

`element` (GstBase.BaseSink) –
No description available
`ringbuffer` (GstAudio.AudioRingBuffer) –
No description available
`buffer_time` (int) –
No description available
`latency_time` (int) –
No description available
`next_sample` (int) –
No description available
`provided_clock` (Gst.Clock) –
No description available
`eos_rendering` (bool) –
No description available

### Methods

#### gst_audio_base_sink_create_ringbuffer

    GstAudioRingBuffer *
    gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)

Create and return the GstAudioRingBuffer for sink. This function will call the ::create_ringbuffer vmethod and will set sink as the parent of the returned buffer (see gst_object_set_parent).

Parameters:

`sink` –

a GstAudioBaseSink.

Returns ( [transfer: none][nullable]) –

The new ringbuffer of sink.

#### GstAudio.AudioBaseSink.prototype.create_ringbuffer

    function GstAudio.AudioBaseSink.prototype.create_ringbuffer(): {
        // javascript wrapper for 'gst_audio_base_sink_create_ringbuffer'
    }

Create and return the GstAudio.AudioRingBuffer for sink. This function will call the ::create_ringbuffer vmethod and will set sink as the parent of the returned buffer (see Gst.Object.prototype.set_parent).

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink.

Returns (GstAudio.AudioRingBuffer) –

The new ringbuffer of sink.

#### GstAudio.AudioBaseSink.create_ringbuffer

    def GstAudio.AudioBaseSink.create_ringbuffer (self):
        #python wrapper for 'gst_audio_base_sink_create_ringbuffer'

Create and return the GstAudio.AudioRingBuffer for sink. This function will call the ::create_ringbuffer vmethod and will set sink as the parent of the returned buffer (see Gst.Object.set_parent).

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink.

Returns (GstAudio.AudioRingBuffer) –

The new ringbuffer of sink.

#### gst_audio_base_sink_get_alignment_threshold

    GstClockTime
    gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)

Get the current alignment threshold, in nanoseconds, used by sink.

Parameters:

`sink` –

a GstAudioBaseSink

Returns –

The current alignment threshold used by sink.

#### GstAudio.AudioBaseSink.prototype.get_alignment_threshold

    function GstAudio.AudioBaseSink.prototype.get_alignment_threshold(): {
        // javascript wrapper for 'gst_audio_base_sink_get_alignment_threshold'
    }

Get the current alignment threshold, in nanoseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (Number) –

The current alignment threshold used by sink.

#### GstAudio.AudioBaseSink.get_alignment_threshold

    def GstAudio.AudioBaseSink.get_alignment_threshold (self):
        #python wrapper for 'gst_audio_base_sink_get_alignment_threshold'

Get the current alignment threshold, in nanoseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (int) –

The current alignment threshold used by sink.

#### gst_audio_base_sink_get_discont_wait

    GstClockTime
    gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)

Get the current discont wait, in nanoseconds, used by sink.

Parameters:

`sink` –

a GstAudioBaseSink

Returns –

The current discont wait used by sink.

#### GstAudio.AudioBaseSink.prototype.get_discont_wait

    function GstAudio.AudioBaseSink.prototype.get_discont_wait(): {
        // javascript wrapper for 'gst_audio_base_sink_get_discont_wait'
    }

Get the current discont wait, in nanoseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (Number) –

The current discont wait used by sink.

#### GstAudio.AudioBaseSink.get_discont_wait

    def GstAudio.AudioBaseSink.get_discont_wait (self):
        #python wrapper for 'gst_audio_base_sink_get_discont_wait'

Get the current discont wait, in nanoseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (int) –

The current discont wait used by sink.

#### gst_audio_base_sink_get_drift_tolerance

    gint64
    gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)

Get the current drift tolerance, in microseconds, used by sink.

Parameters:

`sink` –

a GstAudioBaseSink

Returns –

The current drift tolerance used by sink.

#### GstAudio.AudioBaseSink.prototype.get_drift_tolerance

    function GstAudio.AudioBaseSink.prototype.get_drift_tolerance(): {
        // javascript wrapper for 'gst_audio_base_sink_get_drift_tolerance'
    }

Get the current drift tolerance, in microseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (Number) –

The current drift tolerance used by sink.

#### GstAudio.AudioBaseSink.get_drift_tolerance

    def GstAudio.AudioBaseSink.get_drift_tolerance (self):
        #python wrapper for 'gst_audio_base_sink_get_drift_tolerance'

Get the current drift tolerance, in microseconds, used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (int) –

The current drift tolerance used by sink.

#### gst_audio_base_sink_get_provide_clock

    gboolean
    gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)

Queries whether sink will provide a clock or not. See also gst_audio_base_sink_set_provide_clock.

Parameters:

`sink` –

a GstAudioBaseSink

Returns –

TRUE if sink will provide a clock.

#### GstAudio.AudioBaseSink.prototype.get_provide_clock

    function GstAudio.AudioBaseSink.prototype.get_provide_clock(): {
        // javascript wrapper for 'gst_audio_base_sink_get_provide_clock'
    }

Queries whether sink will provide a clock or not. See also gst_audio_base_sink_set_provide_clock.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (Number) –

TRUE if sink will provide a clock.

#### GstAudio.AudioBaseSink.get_provide_clock

    def GstAudio.AudioBaseSink.get_provide_clock (self):
        #python wrapper for 'gst_audio_base_sink_get_provide_clock'

Queries whether sink will provide a clock or not. See also gst_audio_base_sink_set_provide_clock.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (bool) –

TRUE if sink will provide a clock.

#### gst_audio_base_sink_get_slave_method

    GstAudioBaseSinkSlaveMethod
    gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)

Get the current slave method used by sink.

Parameters:

`sink` –

a GstAudioBaseSink

Returns –

The current slave method used by sink.

#### GstAudio.AudioBaseSink.prototype.get_slave_method

    function GstAudio.AudioBaseSink.prototype.get_slave_method(): {
        // javascript wrapper for 'gst_audio_base_sink_get_slave_method'
    }

Get the current slave method used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (GstAudio.AudioBaseSinkSlaveMethod) –

The current slave method used by sink.

#### GstAudio.AudioBaseSink.get_slave_method

    def GstAudio.AudioBaseSink.get_slave_method (self):
        #python wrapper for 'gst_audio_base_sink_get_slave_method'

Get the current slave method used by sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Returns (GstAudio.AudioBaseSinkSlaveMethod) –

The current slave method used by sink.

#### gst_audio_base_sink_report_device_failure

    gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink)

Informs this base class that the audio output device has failed for some reason, causing a discontinuity (for example, because the device recovered from the error, but lost all contents of its ring buffer). This function is typically called by derived classes, and is useful for the custom slave method.

Parameters:

`sink` –

a GstAudioBaseSink

Since : 1.6

#### GstAudio.AudioBaseSink.prototype.report_device_failure

    function GstAudio.AudioBaseSink.prototype.report_device_failure(): {
        // javascript wrapper for 'gst_audio_base_sink_report_device_failure'
    }

Informs this base class that the audio output device has failed for some reason, causing a discontinuity (for example, because the device recovered from the error, but lost all contents of its ring buffer). This function is typically called by derived classes, and is useful for the custom slave method.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Since : 1.6

#### GstAudio.AudioBaseSink.report_device_failure

    def GstAudio.AudioBaseSink.report_device_failure (self):
        #python wrapper for 'gst_audio_base_sink_report_device_failure'

Informs this base class that the audio output device has failed for some reason, causing a discontinuity (for example, because the device recovered from the error, but lost all contents of its ring buffer). This function is typically called by derived classes, and is useful for the custom slave method.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

Since : 1.6

#### gst_audio_base_sink_set_alignment_threshold

    gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
                                                 GstClockTime alignment_threshold)

Controls the sink's alignment threshold.

Parameters:

`sink` –

a GstAudioBaseSink

`alignment_threshold` –

the new alignment threshold in nanoseconds

#### GstAudio.AudioBaseSink.prototype.set_alignment_threshold

    function GstAudio.AudioBaseSink.prototype.set_alignment_threshold(alignment_threshold: Number): {
        // javascript wrapper for 'gst_audio_base_sink_set_alignment_threshold'
    }

Controls the sink's alignment threshold.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`alignment_threshold` (Number) –

the new alignment threshold in nanoseconds

#### GstAudio.AudioBaseSink.set_alignment_threshold

    def GstAudio.AudioBaseSink.set_alignment_threshold (self, alignment_threshold):
        #python wrapper for 'gst_audio_base_sink_set_alignment_threshold'

Controls the sink's alignment threshold.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`alignment_threshold` (int) –

the new alignment threshold in nanoseconds

#### gst_audio_base_sink_set_custom_slaving_callback

    gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink,
                                                     GstAudioBaseSinkCustomSlavingCallback callback,
                                                     gpointer user_data,
                                                     GDestroyNotify notify)

Sets the custom slaving callback. This callback will be invoked if the slave-method property is set to GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink receives and plays samples.

Setting the callback to NULL causes the sink to behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE method were used.

Parameters:

`sink` –

a GstAudioBaseSink

`callback` –

a GstAudioBaseSinkCustomSlavingCallback

`user_data` –

user data passed to the callback

`notify` –

called when user_data becomes unused

Since : 1.6

#### GstAudio.AudioBaseSink.prototype.set_custom_slaving_callback

    function GstAudio.AudioBaseSink.prototype.set_custom_slaving_callback(callback: GstAudio.AudioBaseSinkCustomSlavingCallback, user_data: Object): {
        // javascript wrapper for 'gst_audio_base_sink_set_custom_slaving_callback'
    }

Sets the custom slaving callback. This callback will be invoked if the slave-method property is set to GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink receives and plays samples.

Setting the callback to NULL causes the sink to behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE method were used.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`callback` (GstAudio.AudioBaseSinkCustomSlavingCallback) –

a GstAudio.AudioBaseSinkCustomSlavingCallback

`user_data` (Object) –

user data passed to the callback

Since : 1.6

#### GstAudio.AudioBaseSink.set_custom_slaving_callback

    def GstAudio.AudioBaseSink.set_custom_slaving_callback (self, callback, *user_data):
        #python wrapper for 'gst_audio_base_sink_set_custom_slaving_callback'

Sets the custom slaving callback. This callback will be invoked if the slave-method property is set to GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink receives and plays samples.

Setting the callback to NULL causes the sink to behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE method were used.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`callback` (GstAudio.AudioBaseSinkCustomSlavingCallback) –

a GstAudio.AudioBaseSinkCustomSlavingCallback

`user_data` (variadic) –

user data passed to the callback

Since : 1.6

#### gst_audio_base_sink_set_discont_wait

    gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
                                          GstClockTime discont_wait)

Controls how long the sink will wait before creating a discontinuity.

Parameters:

`sink` –

a GstAudioBaseSink

`discont_wait` –

the new discont wait in nanoseconds

#### GstAudio.AudioBaseSink.prototype.set_discont_wait

    function GstAudio.AudioBaseSink.prototype.set_discont_wait(discont_wait: Number): {
        // javascript wrapper for 'gst_audio_base_sink_set_discont_wait'
    }

Controls how long the sink will wait before creating a discontinuity.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`discont_wait` (Number) –

the new discont wait in nanoseconds

#### GstAudio.AudioBaseSink.set_discont_wait

    def GstAudio.AudioBaseSink.set_discont_wait (self, discont_wait):
        #python wrapper for 'gst_audio_base_sink_set_discont_wait'

Controls how long the sink will wait before creating a discontinuity.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`discont_wait` (int) –

the new discont wait in nanoseconds

#### gst_audio_base_sink_set_drift_tolerance

    gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
                                             gint64 drift_tolerance)

Controls the sink's drift tolerance.

Parameters:

`sink` –

a GstAudioBaseSink

`drift_tolerance` –

the new drift tolerance in microseconds

#### GstAudio.AudioBaseSink.prototype.set_drift_tolerance

    function GstAudio.AudioBaseSink.prototype.set_drift_tolerance(drift_tolerance: Number): {
        // javascript wrapper for 'gst_audio_base_sink_set_drift_tolerance'
    }

Controls the sink's drift tolerance.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`drift_tolerance` (Number) –

the new drift tolerance in microseconds

#### GstAudio.AudioBaseSink.set_drift_tolerance

    def GstAudio.AudioBaseSink.set_drift_tolerance (self, drift_tolerance):
        #python wrapper for 'gst_audio_base_sink_set_drift_tolerance'

Controls the sink's drift tolerance.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`drift_tolerance` (int) –

the new drift tolerance in microseconds

#### gst_audio_base_sink_set_provide_clock

    gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
                                           gboolean provide)

Controls whether sink will provide a clock or not. If provide is TRUE, gst_element_provide_clock will return a clock that reflects the datarate of sink. If provide is FALSE, gst_element_provide_clock will return NULL.

Parameters:

`sink` –

a GstAudioBaseSink

`provide` –

new state

#### GstAudio.AudioBaseSink.prototype.set_provide_clock

    function GstAudio.AudioBaseSink.prototype.set_provide_clock(provide: Number): {
        // javascript wrapper for 'gst_audio_base_sink_set_provide_clock'
    }

Controls whether sink will provide a clock or not. If provide is TRUE, Gst.Element.prototype.provide_clock will return a clock that reflects the datarate of sink. If provide is FALSE, Gst.Element.prototype.provide_clock will return NULL.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`provide` (Number) –

new state

#### GstAudio.AudioBaseSink.set_provide_clock

    def GstAudio.AudioBaseSink.set_provide_clock (self, provide):
        #python wrapper for 'gst_audio_base_sink_set_provide_clock'

Controls whether sink will provide a clock or not. If provide is TRUE, Gst.Element.provide_clock will return a clock that reflects the datarate of sink. If provide is FALSE, Gst.Element.provide_clock will return NULL.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`provide` (bool) –

new state

#### gst_audio_base_sink_set_slave_method

    gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
                                          GstAudioBaseSinkSlaveMethod method)

Controls how clock slaving will be performed in sink.

Parameters:

`sink` –

a GstAudioBaseSink

`method` –

the new slave method

#### GstAudio.AudioBaseSink.prototype.set_slave_method

    function GstAudio.AudioBaseSink.prototype.set_slave_method(method: GstAudio.AudioBaseSinkSlaveMethod): {
        // javascript wrapper for 'gst_audio_base_sink_set_slave_method'
    }

Controls how clock slaving will be performed in sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`method` (GstAudio.AudioBaseSinkSlaveMethod) –

the new slave method

#### GstAudio.AudioBaseSink.set_slave_method

    def GstAudio.AudioBaseSink.set_slave_method (self, method):
        #python wrapper for 'gst_audio_base_sink_set_slave_method'

Controls how clock slaving will be performed in sink.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`method` (GstAudio.AudioBaseSinkSlaveMethod) –

the new slave method

### Properties

#### alignment-threshold

    “alignment-threshold” guint64

Flags : Read / Write

#### alignment-threshold

    “alignment-threshold” Number

Flags : Read / Write

#### alignment_threshold

    “self.props.alignment_threshold” int

Flags : Read / Write

#### buffer-time

    “buffer-time” gint64

Flags : Read / Write

#### buffer-time

    “buffer-time” Number

Flags : Read / Write

#### buffer_time

    “self.props.buffer_time” int

Flags : Read / Write

#### can-activate-pull

    “can-activate-pull” gboolean

Flags : Read / Write

#### can-activate-pull

    “can-activate-pull” Number

Flags : Read / Write

#### can_activate_pull

    “self.props.can_activate_pull” bool

Flags : Read / Write

#### discont-wait

    “discont-wait” guint64

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

Flags : Read / Write

#### discont-wait

    “discont-wait” Number

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

Flags : Read / Write

#### discont_wait

    “self.props.discont_wait” int

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

Flags : Read / Write

#### drift-tolerance

    “drift-tolerance” gint64

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

Flags : Read / Write

#### drift-tolerance

    “drift-tolerance” Number

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

Flags : Read / Write

#### drift_tolerance

    “self.props.drift_tolerance” int

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

Flags : Read / Write

#### latency-time

    “latency-time” gint64

Flags : Read / Write

#### latency-time

    “latency-time” Number

Flags : Read / Write

#### latency_time

    “self.props.latency_time” int

Flags : Read / Write

#### provide-clock

    “provide-clock” gboolean

Flags : Read / Write

#### provide-clock

    “provide-clock” Number

Flags : Read / Write

#### provide_clock

    “self.props.provide_clock” bool

Flags : Read / Write

#### slave-method

    “slave-method” GstAudioBaseSinkSlaveMethod *

Flags : Read / Write

#### slave-method

    “slave-method” GstAudio.AudioBaseSinkSlaveMethod

Flags : Read / Write

#### slave_method

    “self.props.slave_method” GstAudio.AudioBaseSinkSlaveMethod

Flags : Read / Write

### Virtual Methods

#### create_ringbuffer

    GstAudioRingBuffer *
    create_ringbuffer (GstAudioBaseSink * sink)

create and return a GstAudioRingBuffer to write to.

Parameters:

`sink` –
No description available
Returns –
No description available

#### vfunc_create_ringbuffer

    function vfunc_create_ringbuffer(sink: GstAudio.AudioBaseSink): {
        // javascript implementation of the 'create_ringbuffer' virtual method
    }

create and return a GstAudio.AudioRingBuffer to write to.

Parameters:

`sink` (GstAudio.AudioBaseSink) –
No description available
Returns (GstAudio.AudioRingBuffer) –
No description available

#### do_create_ringbuffer

    def do_create_ringbuffer (sink):
        #python implementation of the 'create_ringbuffer' virtual method

create and return a GstAudio.AudioRingBuffer to write to.

Parameters:

`sink` (GstAudio.AudioBaseSink) –
No description available
Returns (GstAudio.AudioRingBuffer) –
No description available

#### payload

    GstBuffer *
    payload (GstAudioBaseSink * sink,
             GstBuffer * buffer)

payload data in a format suitable to write to the sink. If no payloading is required, returns a reffed copy of the original buffer, else returns the payloaded buffer with all other metadata copied.

Parameters:

`sink` –
No description available
`buffer` –
No description available
Returns –
No description available

#### vfunc_payload

    function vfunc_payload(sink: GstAudio.AudioBaseSink, buffer: Gst.Buffer): {
        // javascript implementation of the 'payload' virtual method
    }

payload data in a format suitable to write to the sink. If no payloading is required, returns a reffed copy of the original buffer, else returns the payloaded buffer with all other metadata copied.

Parameters:

`sink` (GstAudio.AudioBaseSink) –
No description available
`buffer` (Gst.Buffer) –
No description available
Returns (Gst.Buffer) –
No description available

#### do_payload

    def do_payload (sink, buffer):
        #python implementation of the 'payload' virtual method

payload data in a format suitable to write to the sink. If no payloading is required, returns a reffed copy of the original buffer, else returns the payloaded buffer with all other metadata copied.

Parameters:

`sink` (GstAudio.AudioBaseSink) –
No description available
`buffer` (Gst.Buffer) –
No description available
Returns (Gst.Buffer) –
No description available

## Function Macros

### GST_AUDIO_BASE_SINK_CAST

    #define GST_AUDIO_BASE_SINK_CAST(obj)           ((GstAudioBaseSink*)obj)

### GST_AUDIO_BASE_SINK_CLOCK

    #define GST_AUDIO_BASE_SINK_CLOCK(obj)   (GST_AUDIO_BASE_SINK (obj)->clock)

Get the GstClock of obj.

Parameters:

`obj` –

a GstAudioBaseSink

### GST_AUDIO_BASE_SINK_PAD

    #define GST_AUDIO_BASE_SINK_PAD(obj)     (GST_BASE_SINK (obj)->sinkpad)

Get the sink GstPad of obj.

Parameters:

`obj` –

a GstAudioBaseSink

## Enumerations

### GstAudioBaseSinkDiscontReason

Different possible reasons for discontinuities. This enum is useful for the custom slave method.

##### Members

`GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT` (0) –

No discontinuity occurred

`GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS` (1) –

New caps are set, causing renegotiotion

`GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH` (2) –

Samples have been flushed

`GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY` (3) –

Sink was synchronized to the estimated latency (occurs during initialization)

`GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT` (4) –

Aligning buffers failed because the timestamps are too discontinuous

`GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE` (5) –

Audio output device experienced and recovered from an error but introduced latency in the process (see also gst_audio_base_sink_report_device_failure)

Since : 1.6

### GstAudio.AudioBaseSinkDiscontReason

Different possible reasons for discontinuities. This enum is useful for the custom slave method.

##### Members

`GstAudio.AudioBaseSinkDiscontReason.NO_DISCONT` (0) –

No discontinuity occurred

`GstAudio.AudioBaseSinkDiscontReason.NEW_CAPS` (1) –

New caps are set, causing renegotiotion

`GstAudio.AudioBaseSinkDiscontReason.FLUSH` (2) –

Samples have been flushed

`GstAudio.AudioBaseSinkDiscontReason.SYNC_LATENCY` (3) –

Sink was synchronized to the estimated latency (occurs during initialization)

`GstAudio.AudioBaseSinkDiscontReason.ALIGNMENT` (4) –

Aligning buffers failed because the timestamps are too discontinuous

`GstAudio.AudioBaseSinkDiscontReason.DEVICE_FAILURE` (5) –

Audio output device experienced and recovered from an error but introduced latency in the process (see also GstAudio.AudioBaseSink.prototype.report_device_failure)

Since : 1.6

### GstAudio.AudioBaseSinkDiscontReason

Different possible reasons for discontinuities. This enum is useful for the custom slave method.

##### Members

`GstAudio.AudioBaseSinkDiscontReason.NO_DISCONT` (0) –

No discontinuity occurred

`GstAudio.AudioBaseSinkDiscontReason.NEW_CAPS` (1) –

New caps are set, causing renegotiotion

`GstAudio.AudioBaseSinkDiscontReason.FLUSH` (2) –

Samples have been flushed

`GstAudio.AudioBaseSinkDiscontReason.SYNC_LATENCY` (3) –

Sink was synchronized to the estimated latency (occurs during initialization)

`GstAudio.AudioBaseSinkDiscontReason.ALIGNMENT` (4) –

Aligning buffers failed because the timestamps are too discontinuous

`GstAudio.AudioBaseSinkDiscontReason.DEVICE_FAILURE` (5) –

Audio output device experienced and recovered from an error but introduced latency in the process (see also GstAudio.AudioBaseSink.report_device_failure)

Since : 1.6

### GstAudioBaseSinkSlaveMethod

Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.

##### Members

`GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE` (0) –

Resample to match the master clock

`GST_AUDIO_BASE_SINK_SLAVE_SKEW` (1) –

Adjust playout pointer when master clock drifts too much.

`GST_AUDIO_BASE_SINK_SLAVE_NONE` (2) –

No adjustment is done.

`GST_AUDIO_BASE_SINK_SLAVE_CUSTOM` (3) –

Use custom clock slaving algorithm (Since: 1.6)

### GstAudio.AudioBaseSinkSlaveMethod

Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.

##### Members

`GstAudio.AudioBaseSinkSlaveMethod.RESAMPLE` (0) –

Resample to match the master clock

`GstAudio.AudioBaseSinkSlaveMethod.SKEW` (1) –

Adjust playout pointer when master clock drifts too much.

`GstAudio.AudioBaseSinkSlaveMethod.NONE` (2) –

No adjustment is done.

`GstAudio.AudioBaseSinkSlaveMethod.CUSTOM` (3) –

Use custom clock slaving algorithm (Since: 1.6)

### GstAudio.AudioBaseSinkSlaveMethod

Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.

##### Members

`GstAudio.AudioBaseSinkSlaveMethod.RESAMPLE` (0) –

Resample to match the master clock

`GstAudio.AudioBaseSinkSlaveMethod.SKEW` (1) –

Adjust playout pointer when master clock drifts too much.

`GstAudio.AudioBaseSinkSlaveMethod.NONE` (2) –

No adjustment is done.

`GstAudio.AudioBaseSinkSlaveMethod.CUSTOM` (3) –

Use custom clock slaving algorithm (Since: 1.6)

## Callbacks

### GstAudioBaseSinkCustomSlavingCallback

    (*GstAudioBaseSinkCustomSlavingCallback) (GstAudioBaseSink * sink,
                                              GstClockTime etime,
                                              GstClockTime itime,
                                              GstClockTimeDiff * requested_skew,
                                              GstAudioBaseSinkDiscontReason discont_reason,
                                              gpointer user_data)

This function is set with gst_audio_base_sink_set_custom_slaving_callback and is called during playback. It receives the current time of external and internal clocks, which the callback can then use to apply any custom slaving/synchronization schemes.

The external clock is the sink's element clock, the internal one is the internal audio clock. The internal audio clock's calibration is applied to the timestamps before they are passed to the callback. The difference between etime and itime is the skew; how much internal and external clock lie apart from each other. A skew of 0 means both clocks are perfectly in sync. itime > etime means the external clock is going slower, while itime < etime means it is going faster than the internal clock. etime and itime are always valid timestamps, except for when a discontinuity happens.

requested_skew is an output value the callback can write to. It informs the sink of whether or not it should move the playout pointer, and if so, by how much. This pointer is only NULL if a discontinuity occurs; otherwise, it is safe to write to *requested_skew. The default skew is 0.

The sink may experience discontinuities. If one happens, discont is TRUE, itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. This makes it possible to reset custom clock slaving algorithms when a discontinuity happens.

Parameters:

`sink` –

a GstAudioBaseSink

`etime` –

external clock time

`itime` –

internal clock time

`requested_skew` –

skew amount requested by the callback

`discont_reason` –

reason for discontinuity (if any)

`user_data` –

user data

Since : 1.6

### GstAudio.AudioBaseSinkCustomSlavingCallback

    function GstAudio.AudioBaseSinkCustomSlavingCallback(sink: GstAudio.AudioBaseSink, etime: Number, itime: Number, requested_skew: Number, discont_reason: GstAudio.AudioBaseSinkDiscontReason, user_data: Object): {
        // javascript wrapper for 'GstAudioBaseSinkCustomSlavingCallback'
    }

This function is set with GstAudio.AudioBaseSink.prototype.set_custom_slaving_callback and is called during playback. It receives the current time of external and internal clocks, which the callback can then use to apply any custom slaving/synchronization schemes.

The external clock is the sink's element clock, the internal one is the internal audio clock. The internal audio clock's calibration is applied to the timestamps before they are passed to the callback. The difference between etime and itime is the skew; how much internal and external clock lie apart from each other. A skew of 0 means both clocks are perfectly in sync. itime > etime means the external clock is going slower, while itime < etime means it is going faster than the internal clock. etime and itime are always valid timestamps, except for when a discontinuity happens.

requested_skew is an output value the callback can write to. It informs the sink of whether or not it should move the playout pointer, and if so, by how much. This pointer is only NULL if a discontinuity occurs; otherwise, it is safe to write to *requested_skew. The default skew is 0.

The sink may experience discontinuities. If one happens, discont is TRUE, itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. This makes it possible to reset custom clock slaving algorithms when a discontinuity happens.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`etime` (Number) –

external clock time

`itime` (Number) –

internal clock time

`requested_skew` (Number) –

skew amount requested by the callback

`discont_reason` (GstAudio.AudioBaseSinkDiscontReason) –

reason for discontinuity (if any)

`user_data` (Object) –

user data

Since : 1.6

### GstAudio.AudioBaseSinkCustomSlavingCallback

    def GstAudio.AudioBaseSinkCustomSlavingCallback (sink, etime, itime, requested_skew, discont_reason, *user_data):
        #python wrapper for 'GstAudioBaseSinkCustomSlavingCallback'

This function is set with GstAudio.AudioBaseSink.set_custom_slaving_callback and is called during playback. It receives the current time of external and internal clocks, which the callback can then use to apply any custom slaving/synchronization schemes.

The external clock is the sink's element clock, the internal one is the internal audio clock. The internal audio clock's calibration is applied to the timestamps before they are passed to the callback. The difference between etime and itime is the skew; how much internal and external clock lie apart from each other. A skew of 0 means both clocks are perfectly in sync. itime > etime means the external clock is going slower, while itime < etime means it is going faster than the internal clock. etime and itime are always valid timestamps, except for when a discontinuity happens.

requested_skew is an output value the callback can write to. It informs the sink of whether or not it should move the playout pointer, and if so, by how much. This pointer is only NULL if a discontinuity occurs; otherwise, it is safe to write to *requested_skew. The default skew is 0.

The sink may experience discontinuities. If one happens, discont is TRUE, itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. This makes it possible to reset custom clock slaving algorithms when a discontinuity happens.

Parameters:

`sink` (GstAudio.AudioBaseSink) –

a GstAudio.AudioBaseSink

`etime` (int) –

external clock time

`itime` (int) –

internal clock time

`requested_skew` (int) –

skew amount requested by the callback

`discont_reason` (GstAudio.AudioBaseSinkDiscontReason) –

reason for discontinuity (if any)

`user_data` (variadic) –

user data

Since : 1.6

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