# GstAudioRingBuffer

This object is the base class for audio ringbuffers used by the base audio source and sink classes.

The ringbuffer abstracts a circular buffer of data. One reader and one writer can operate on the data from different threads in a lockfree manner. The base class is sufficiently flexible to be used as an abstraction for DMA based ringbuffers as well as a pure software implementations.

## GstAudioRingBuffer

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstAudioRingBuffer

The ringbuffer base class structure.

### Members

`object` (GstObject) –
No description available
`cond` (GCond) –

used to signal start/stop/pause/resume actions

`open` (gboolean) –

boolean indicating that the ringbuffer is open

`acquired` (gboolean) –

boolean indicating that the ringbuffer is acquired

`memory` (guint8*) –

data in the ringbuffer

`size` (gsize) –

size of data in the ringbuffer

`spec` (GstAudioRingBufferSpec) –

format and layout of the ringbuffer data

`samples_per_seg` (gint) –

number of samples in one segment

`empty_seg` (guint8*) –

pointer to memory holding one segment of silence samples

`state` (gint) –

state of the buffer

`segdone` (gint) –

readpointer in the ringbuffer

`segbase` (gint) –

segment corresponding to segment 0 (unused)

`waiting` (gint) –

is a reader or writer waiting for a free segment

### Class structure

#### GstAudioRingBufferClass

The vmethods that subclasses can override to implement the ringbuffer.

##### Fields

`parent_class` (GstObjectClass) –

parent class

### GstAudio.AudioRingBufferClass

The vmethods that subclasses can override to implement the ringbuffer.

##### Attributes

`parent_class` (Gst.ObjectClass) –

parent class

### GstAudio.AudioRingBufferClass

The vmethods that subclasses can override to implement the ringbuffer.

##### Attributes

`parent_class` (Gst.ObjectClass) –

parent class

## GstAudio.AudioRingBuffer

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstAudio.AudioRingBuffer

The ringbuffer base class structure.

### Members

`object` (Gst.Object) –
No description available
`cond` (GLib.Cond) –

used to signal start/stop/pause/resume actions

`open` (Number) –

boolean indicating that the ringbuffer is open

`acquired` (Number) –

boolean indicating that the ringbuffer is acquired

`memory` (Number) –

data in the ringbuffer

`size` (Number) –

size of data in the ringbuffer

`spec` (GstAudio.AudioRingBufferSpec) –

format and layout of the ringbuffer data

`samples_per_seg` (Number) –

number of samples in one segment

`empty_seg` (Number) –

pointer to memory holding one segment of silence samples

`state` (Number) –

state of the buffer

`segdone` (Number) –

readpointer in the ringbuffer

`segbase` (Number) –

segment corresponding to segment 0 (unused)

`waiting` (Number) –

is a reader or writer waiting for a free segment

## GstAudio.AudioRingBuffer

    GObject.Object
        ╰──GObject.InitiallyUnowned
            ╰──Gst.Object
                ╰──GstAudio.AudioRingBuffer

The ringbuffer base class structure.

### Members

`object` (Gst.Object) –
No description available
`cond` (GLib.Cond) –

used to signal start/stop/pause/resume actions

`open` (bool) –

boolean indicating that the ringbuffer is open

`acquired` (bool) –

boolean indicating that the ringbuffer is acquired

`memory` (int) –

data in the ringbuffer

`size` (int) –

size of data in the ringbuffer

`spec` (GstAudio.AudioRingBufferSpec) –

format and layout of the ringbuffer data

`samples_per_seg` (int) –

number of samples in one segment

`empty_seg` (int) –

pointer to memory holding one segment of silence samples

`state` (int) –

state of the buffer

`segdone` (int) –

readpointer in the ringbuffer

`segbase` (int) –

segment corresponding to segment 0 (unused)

`waiting` (int) –

is a reader or writer waiting for a free segment

### Methods

#### gst_audio_ring_buffer_acquire

    gboolean
    gst_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
                                   GstAudioRingBufferSpec * spec)

Allocate the resources for the ringbuffer. This function fills in the data pointer of the ring buffer with a valid GstBuffer to which samples can be written.

Parameters:

`buf` –

the GstAudioRingBuffer to acquire

`spec` –

the specs of the buffer

Returns –

TRUE if the device could be acquired, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.acquire

    function GstAudio.AudioRingBuffer.prototype.acquire(spec: GstAudio.AudioRingBufferSpec): {
        // javascript wrapper for 'gst_audio_ring_buffer_acquire'
    }

Allocate the resources for the ringbuffer. This function fills in the data pointer of the ring buffer with a valid Gst.Buffer to which samples can be written.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to acquire

`spec` (GstAudio.AudioRingBufferSpec) –

the specs of the buffer

Returns (Number) –

TRUE if the device could be acquired, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.acquire

    def GstAudio.AudioRingBuffer.acquire (self, spec):
        #python wrapper for 'gst_audio_ring_buffer_acquire'

Allocate the resources for the ringbuffer. This function fills in the data pointer of the ring buffer with a valid Gst.Buffer to which samples can be written.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to acquire

`spec` (GstAudio.AudioRingBufferSpec) –

the specs of the buffer

Returns (bool) –

TRUE if the device could be acquired, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_activate

    gboolean
    gst_audio_ring_buffer_activate (GstAudioRingBuffer * buf,
                                    gboolean active)

Activate buf to start or stop pulling data.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to activate

`active` –

the new mode

Returns –

TRUE if the device could be activated in the requested mode, FALSE on error.

#### GstAudio.AudioRingBuffer.prototype.activate

    function GstAudio.AudioRingBuffer.prototype.activate(active: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_activate'
    }

Activate buf to start or stop pulling data.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to activate

`active` (Number) –

the new mode

Returns (Number) –

TRUE if the device could be activated in the requested mode, FALSE on error.

#### GstAudio.AudioRingBuffer.activate

    def GstAudio.AudioRingBuffer.activate (self, active):
        #python wrapper for 'gst_audio_ring_buffer_activate'

Activate buf to start or stop pulling data.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to activate

`active` (bool) –

the new mode

Returns (bool) –

TRUE if the device could be activated in the requested mode, FALSE on error.

#### gst_audio_ring_buffer_advance

    gst_audio_ring_buffer_advance (GstAudioRingBuffer * buf,
                                   guint advance)

Subclasses should call this function to notify the fact that advance segments are now processed by the device.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to advance

`advance` –

the number of segments written

#### GstAudio.AudioRingBuffer.prototype.advance

    function GstAudio.AudioRingBuffer.prototype.advance(advance: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_advance'
    }

Subclasses should call this function to notify the fact that advance segments are now processed by the device.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to advance

`advance` (Number) –

the number of segments written

#### GstAudio.AudioRingBuffer.advance

    def GstAudio.AudioRingBuffer.advance (self, advance):
        #python wrapper for 'gst_audio_ring_buffer_advance'

Subclasses should call this function to notify the fact that advance segments are now processed by the device.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to advance

`advance` (int) –

the number of segments written

#### gst_audio_ring_buffer_clear

    gst_audio_ring_buffer_clear (GstAudioRingBuffer * buf,
                                 gint segment)

Clear the given segment of the buffer with silence samples. This function is used by subclasses.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to clear

`segment` –

the segment to clear

#### GstAudio.AudioRingBuffer.prototype.clear

    function GstAudio.AudioRingBuffer.prototype.clear(segment: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_clear'
    }

Clear the given segment of the buffer with silence samples. This function is used by subclasses.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to clear

`segment` (Number) –

the segment to clear

#### GstAudio.AudioRingBuffer.clear

    def GstAudio.AudioRingBuffer.clear (self, segment):
        #python wrapper for 'gst_audio_ring_buffer_clear'

Clear the given segment of the buffer with silence samples. This function is used by subclasses.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to clear

`segment` (int) –

the segment to clear

#### gst_audio_ring_buffer_clear_all

    gst_audio_ring_buffer_clear_all (GstAudioRingBuffer * buf)

Clear all samples from the ringbuffer.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to clear

#### GstAudio.AudioRingBuffer.prototype.clear_all

    function GstAudio.AudioRingBuffer.prototype.clear_all(): {
        // javascript wrapper for 'gst_audio_ring_buffer_clear_all'
    }

Clear all samples from the ringbuffer.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to clear

#### GstAudio.AudioRingBuffer.clear_all

    def GstAudio.AudioRingBuffer.clear_all (self):
        #python wrapper for 'gst_audio_ring_buffer_clear_all'

Clear all samples from the ringbuffer.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to clear

#### gst_audio_ring_buffer_close_device

    gboolean
    gst_audio_ring_buffer_close_device (GstAudioRingBuffer * buf)

Close the audio device associated with the ring buffer. The ring buffer should already have been released via gst_audio_ring_buffer_release.

Parameters:

`buf` –

the GstAudioRingBuffer

Returns –

TRUE if the device could be closed, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.close_device

    function GstAudio.AudioRingBuffer.prototype.close_device(): {
        // javascript wrapper for 'gst_audio_ring_buffer_close_device'
    }

Close the audio device associated with the ring buffer. The ring buffer should already have been released via GstAudio.AudioRingBuffer.prototype.release.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (Number) –

TRUE if the device could be closed, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.close_device

    def GstAudio.AudioRingBuffer.close_device (self):
        #python wrapper for 'gst_audio_ring_buffer_close_device'

Close the audio device associated with the ring buffer. The ring buffer should already have been released via GstAudio.AudioRingBuffer.release.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (bool) –

TRUE if the device could be closed, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_commit

    guint
    gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf,
                                  guint64* sample,
                                  guint8* data,
                                  gint in_samples,
                                  gint out_samples,
                                  gint* accum)

Commit in_samples samples pointed to by data to the ringbuffer buf.

in_samples and out_samples define the rate conversion to perform on the samples in data. For negative rates, out_samples must be negative and in_samples positive.

When out_samples is positive, the first sample will be written at position sample in the ringbuffer. When out_samples is negative, the last sample will be written to sample in reverse order.

out_samples does not need to be a multiple of the segment size of the ringbuffer although it is recommended for optimal performance.

accum will hold a temporary accumulator used in rate conversion and should be set to 0 when this function is first called. In case the commit operation is interrupted, one can resume the processing by passing the previously returned accum value back to this function.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to commit

`sample` ( [inout]) –

the sample position of the data

`data` ( [arraylength=in_samples]) –

the data to commit

`in_samples` –

the number of samples in the data to commit

`out_samples` –

the number of samples to write to the ringbuffer

`accum` ( [inout]) –

accumulator for rate conversion.

Returns –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

#### GstAudio.AudioRingBuffer.prototype.commit

    function GstAudio.AudioRingBuffer.prototype.commit(sample: Number, data: [ Number ], in_samples: Number, out_samples: Number, accum: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_commit'
    }

Commit in_samples samples pointed to by data to the ringbuffer buf.

in_samples and out_samples define the rate conversion to perform on the samples in data. For negative rates, out_samples must be negative and in_samples positive.

When out_samples is positive, the first sample will be written at position sample in the ringbuffer. When out_samples is negative, the last sample will be written to sample in reverse order.

out_samples does not need to be a multiple of the segment size of the ringbuffer although it is recommended for optimal performance.

accum will hold a temporary accumulator used in rate conversion and should be set to 0 when this function is first called. In case the commit operation is interrupted, one can resume the processing by passing the previously returned accum value back to this function.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to commit

`sample` (Number) –

the sample position of the data

`data` ([ Number ]) –

the data to commit

`in_samples` (Number) –

the number of samples in the data to commit

`out_samples` (Number) –

the number of samples to write to the ringbuffer

`accum` (Number) –

accumulator for rate conversion.

Returns a tuple made of:

(Number ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

`sample` (Number ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

`accum` (Number ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

#### GstAudio.AudioRingBuffer.commit

    def GstAudio.AudioRingBuffer.commit (self, sample, data, in_samples, out_samples, accum):
        #python wrapper for 'gst_audio_ring_buffer_commit'

Commit in_samples samples pointed to by data to the ringbuffer buf.

in_samples and out_samples define the rate conversion to perform on the samples in data. For negative rates, out_samples must be negative and in_samples positive.

When out_samples is positive, the first sample will be written at position sample in the ringbuffer. When out_samples is negative, the last sample will be written to sample in reverse order.

out_samples does not need to be a multiple of the segment size of the ringbuffer although it is recommended for optimal performance.

accum will hold a temporary accumulator used in rate conversion and should be set to 0 when this function is first called. In case the commit operation is interrupted, one can resume the processing by passing the previously returned accum value back to this function.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to commit

`sample` (int) –

the sample position of the data

`data` ([ int ]) –

the data to commit

`in_samples` (int) –

the number of samples in the data to commit

`out_samples` (int) –

the number of samples to write to the ringbuffer

`accum` (int) –

accumulator for rate conversion.

Returns a tuple made of:

(int ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

`sample` (int ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

`accum` (int ) –

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.

#### gst_audio_ring_buffer_convert

    gboolean
    gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf,
                                   GstFormat src_fmt,
                                   gint64 src_val,
                                   GstFormat dest_fmt,
                                   gint64* dest_val)

Convert src_val in src_fmt to the equivalent value in dest_fmt. The result will be put in dest_val.

Parameters:

`buf` –

the GstAudioRingBuffer

`src_fmt` –

the source format

`src_val` –

the source value

`dest_fmt` –

the destination format

`dest_val` ( [out]) –

a location to store the converted value

Returns –

TRUE if the conversion succeeded.

#### GstAudio.AudioRingBuffer.prototype.convert

    function GstAudio.AudioRingBuffer.prototype.convert(src_fmt: Gst.Format, src_val: Number, dest_fmt: Gst.Format): {
        // javascript wrapper for 'gst_audio_ring_buffer_convert'
    }

Convert src_val in src_fmt to the equivalent value in dest_fmt. The result will be put in dest_val.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`src_fmt` (Gst.Format) –

the source format

`src_val` (Number) –

the source value

`dest_fmt` (Gst.Format) –

the destination format

Returns a tuple made of:

(Number ) –

TRUE if the conversion succeeded.

`dest_val` (Number ) –

TRUE if the conversion succeeded.

#### GstAudio.AudioRingBuffer.convert

    def GstAudio.AudioRingBuffer.convert (self, src_fmt, src_val, dest_fmt):
        #python wrapper for 'gst_audio_ring_buffer_convert'

Convert src_val in src_fmt to the equivalent value in dest_fmt. The result will be put in dest_val.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`src_fmt` (Gst.Format) –

the source format

`src_val` (int) –

the source value

`dest_fmt` (Gst.Format) –

the destination format

Returns a tuple made of:

(bool ) –

TRUE if the conversion succeeded.

`dest_val` (int ) –

TRUE if the conversion succeeded.

#### gst_audio_ring_buffer_delay

    guint
    gst_audio_ring_buffer_delay (GstAudioRingBuffer * buf)

Get the number of samples queued in the audio device. This is usually less than the segment size but can be bigger when the implementation uses another internal buffer between the audio device.

For playback ringbuffers this is the amount of samples transferred from the ringbuffer to the device but still not played.

For capture ringbuffers this is the amount of samples in the device that are not yet transferred to the ringbuffer.

Parameters:

`buf` –

the GstAudioRingBuffer to query

Returns –

The number of samples queued in the audio device.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.delay

    function GstAudio.AudioRingBuffer.prototype.delay(): {
        // javascript wrapper for 'gst_audio_ring_buffer_delay'
    }

Get the number of samples queued in the audio device. This is usually less than the segment size but can be bigger when the implementation uses another internal buffer between the audio device.

For playback ringbuffers this is the amount of samples transferred from the ringbuffer to the device but still not played.

For capture ringbuffers this is the amount of samples in the device that are not yet transferred to the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to query

Returns (Number) –

The number of samples queued in the audio device.

MT safe.

#### GstAudio.AudioRingBuffer.delay

    def GstAudio.AudioRingBuffer.delay (self):
        #python wrapper for 'gst_audio_ring_buffer_delay'

Get the number of samples queued in the audio device. This is usually less than the segment size but can be bigger when the implementation uses another internal buffer between the audio device.

For playback ringbuffers this is the amount of samples transferred from the ringbuffer to the device but still not played.

For capture ringbuffers this is the amount of samples in the device that are not yet transferred to the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to query

Returns (int) –

The number of samples queued in the audio device.

MT safe.

#### gst_audio_ring_buffer_device_is_open

    gboolean
    gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer * buf)

Checks the status of the device associated with the ring buffer.

Parameters:

`buf` –

the GstAudioRingBuffer

Returns –

TRUE if the device was open, FALSE if it was closed.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.device_is_open

    function GstAudio.AudioRingBuffer.prototype.device_is_open(): {
        // javascript wrapper for 'gst_audio_ring_buffer_device_is_open'
    }

Checks the status of the device associated with the ring buffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (Number) –

TRUE if the device was open, FALSE if it was closed.

MT safe.

#### GstAudio.AudioRingBuffer.device_is_open

    def GstAudio.AudioRingBuffer.device_is_open (self):
        #python wrapper for 'gst_audio_ring_buffer_device_is_open'

Checks the status of the device associated with the ring buffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (bool) –

TRUE if the device was open, FALSE if it was closed.

MT safe.

#### gst_audio_ring_buffer_is_acquired

    gboolean
    gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer * buf)

Check if the ringbuffer is acquired and ready to use.

Parameters:

`buf` –

the GstAudioRingBuffer to check

Returns –

TRUE if the ringbuffer is acquired, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.is_acquired

    function GstAudio.AudioRingBuffer.prototype.is_acquired(): {
        // javascript wrapper for 'gst_audio_ring_buffer_is_acquired'
    }

Check if the ringbuffer is acquired and ready to use.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to check

Returns (Number) –

TRUE if the ringbuffer is acquired, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.is_acquired

    def GstAudio.AudioRingBuffer.is_acquired (self):
        #python wrapper for 'gst_audio_ring_buffer_is_acquired'

Check if the ringbuffer is acquired and ready to use.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to check

Returns (bool) –

TRUE if the ringbuffer is acquired, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_is_active

    gboolean
    gst_audio_ring_buffer_is_active (GstAudioRingBuffer * buf)

Check if buf is activated.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer

Returns –

TRUE if the device is active.

#### GstAudio.AudioRingBuffer.prototype.is_active

    function GstAudio.AudioRingBuffer.prototype.is_active(): {
        // javascript wrapper for 'gst_audio_ring_buffer_is_active'
    }

Check if buf is activated.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (Number) –

TRUE if the device is active.

#### GstAudio.AudioRingBuffer.is_active

    def GstAudio.AudioRingBuffer.is_active (self):
        #python wrapper for 'gst_audio_ring_buffer_is_active'

Check if buf is activated.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (bool) –

TRUE if the device is active.

#### gst_audio_ring_buffer_is_flushing

    gboolean
    gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer * buf)

Check if buf is flushing.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer

Returns –

TRUE if the device is flushing.

#### GstAudio.AudioRingBuffer.prototype.is_flushing

    function GstAudio.AudioRingBuffer.prototype.is_flushing(): {
        // javascript wrapper for 'gst_audio_ring_buffer_is_flushing'
    }

Check if buf is flushing.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (Number) –

TRUE if the device is flushing.

#### GstAudio.AudioRingBuffer.is_flushing

    def GstAudio.AudioRingBuffer.is_flushing (self):
        #python wrapper for 'gst_audio_ring_buffer_is_flushing'

Check if buf is flushing.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (bool) –

TRUE if the device is flushing.

#### gst_audio_ring_buffer_may_start

    gst_audio_ring_buffer_may_start (GstAudioRingBuffer * buf,
                                     gboolean allowed)

Tell the ringbuffer that it is allowed to start playback when the ringbuffer is filled with samples.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer

`allowed` –

the new value

#### GstAudio.AudioRingBuffer.prototype.may_start

    function GstAudio.AudioRingBuffer.prototype.may_start(allowed: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_may_start'
    }

Tell the ringbuffer that it is allowed to start playback when the ringbuffer is filled with samples.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`allowed` (Number) –

the new value

#### GstAudio.AudioRingBuffer.may_start

    def GstAudio.AudioRingBuffer.may_start (self, allowed):
        #python wrapper for 'gst_audio_ring_buffer_may_start'

Tell the ringbuffer that it is allowed to start playback when the ringbuffer is filled with samples.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`allowed` (bool) –

the new value

#### gst_audio_ring_buffer_open_device

    gboolean
    gst_audio_ring_buffer_open_device (GstAudioRingBuffer * buf)

Open the audio device associated with the ring buffer. Does not perform any setup on the device. You must open the device before acquiring the ring buffer.

Parameters:

`buf` –

the GstAudioRingBuffer

Returns –

TRUE if the device could be opened, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.open_device

    function GstAudio.AudioRingBuffer.prototype.open_device(): {
        // javascript wrapper for 'gst_audio_ring_buffer_open_device'
    }

Open the audio device associated with the ring buffer. Does not perform any setup on the device. You must open the device before acquiring the ring buffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (Number) –

TRUE if the device could be opened, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.open_device

    def GstAudio.AudioRingBuffer.open_device (self):
        #python wrapper for 'gst_audio_ring_buffer_open_device'

Open the audio device associated with the ring buffer. Does not perform any setup on the device. You must open the device before acquiring the ring buffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

Returns (bool) –

TRUE if the device could be opened, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_pause

    gboolean
    gst_audio_ring_buffer_pause (GstAudioRingBuffer * buf)

Pause processing samples from the ringbuffer.

Parameters:

`buf` –

the GstAudioRingBuffer to pause

Returns –

TRUE if the device could be paused, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.pause

    function GstAudio.AudioRingBuffer.prototype.pause(): {
        // javascript wrapper for 'gst_audio_ring_buffer_pause'
    }

Pause processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to pause

Returns (Number) –

TRUE if the device could be paused, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.pause

    def GstAudio.AudioRingBuffer.pause (self):
        #python wrapper for 'gst_audio_ring_buffer_pause'

Pause processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to pause

Returns (bool) –

TRUE if the device could be paused, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_prepare_read

    gboolean
    gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer * buf,
                                        gint* segment,
                                        guint8** readptr,
                                        gint* len)

Returns a pointer to memory where the data from segment segment can be found. This function is mostly used by subclasses.

Parameters:

`buf` –

the GstAudioRingBuffer to read from

`segment` ( [out]) –

the segment to read

`readptr` –

(out) (array length=len): the pointer to the memory where samples can be read

`len` ( [out]) –

the number of bytes to read

Returns –

FALSE if the buffer is not started.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.prepare_read

    function GstAudio.AudioRingBuffer.prototype.prepare_read(): {
        // javascript wrapper for 'gst_audio_ring_buffer_prepare_read'
    }

Returns a pointer to memory where the data from segment segment can be found. This function is mostly used by subclasses.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to read from

Returns a tuple made of:

(Number ) –

FALSE if the buffer is not started.

MT safe.

`segment` (Number ) –

FALSE if the buffer is not started.

MT safe.

`readptr` ([ Number ] ) –

FALSE if the buffer is not started.

MT safe.

`len` (Number ) –

FALSE if the buffer is not started.

MT safe.

#### GstAudio.AudioRingBuffer.prepare_read

    def GstAudio.AudioRingBuffer.prepare_read (self):
        #python wrapper for 'gst_audio_ring_buffer_prepare_read'

Returns a pointer to memory where the data from segment segment can be found. This function is mostly used by subclasses.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to read from

Returns a tuple made of:

(bool ) –

FALSE if the buffer is not started.

MT safe.

`segment` (int ) –

FALSE if the buffer is not started.

MT safe.

`readptr` ([ int ] ) –

FALSE if the buffer is not started.

MT safe.

`len` (int ) –

FALSE if the buffer is not started.

MT safe.

#### gst_audio_ring_buffer_read

    guint
    gst_audio_ring_buffer_read (GstAudioRingBuffer * buf,
                                guint64 sample,
                                guint8* data,
                                guint len,
                                GstClockTime * timestamp)

Read len samples from the ringbuffer into the memory pointed to by data. The first sample should be read from position sample in the ringbuffer.

len should not be a multiple of the segment size of the ringbuffer although it is recommended.

timestamp will return the timestamp associated with the data returned.

Parameters:

`buf` –

the GstAudioRingBuffer to read from

`sample` –

the sample position of the data

`data` ( [arraylength=len]) –

where the data should be read

`len` –

the number of samples in data to read

`timestamp` ( [out]) –

where the timestamp is returned

Returns –

The number of samples read from the ringbuffer or -1 on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.read

    function GstAudio.AudioRingBuffer.prototype.read(sample: Number, data: [ Number ], len: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_read'
    }

Read len samples from the ringbuffer into the memory pointed to by data. The first sample should be read from position sample in the ringbuffer.

len should not be a multiple of the segment size of the ringbuffer although it is recommended.

timestamp will return the timestamp associated with the data returned.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to read from

`sample` (Number) –

the sample position of the data

`data` ([ Number ]) –

where the data should be read

`len` (Number) –

the number of samples in data to read

Returns a tuple made of:

(Number ) –

The number of samples read from the ringbuffer or -1 on error.

MT safe.

`timestamp` (Number ) –

The number of samples read from the ringbuffer or -1 on error.

MT safe.

#### GstAudio.AudioRingBuffer.read

    def GstAudio.AudioRingBuffer.read (self, sample, data, len):
        #python wrapper for 'gst_audio_ring_buffer_read'

Read len samples from the ringbuffer into the memory pointed to by data. The first sample should be read from position sample in the ringbuffer.

len should not be a multiple of the segment size of the ringbuffer although it is recommended.

timestamp will return the timestamp associated with the data returned.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to read from

`sample` (int) –

the sample position of the data

`data` ([ int ]) –

where the data should be read

`len` (int) –

the number of samples in data to read

Returns a tuple made of:

(int ) –

The number of samples read from the ringbuffer or -1 on error.

MT safe.

`timestamp` (int ) –

The number of samples read from the ringbuffer or -1 on error.

MT safe.

#### gst_audio_ring_buffer_release

    gboolean
    gst_audio_ring_buffer_release (GstAudioRingBuffer * buf)

Free the resources of the ringbuffer.

Parameters:

`buf` –

the GstAudioRingBuffer to release

Returns –

TRUE if the device could be released, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.release

    function GstAudio.AudioRingBuffer.prototype.release(): {
        // javascript wrapper for 'gst_audio_ring_buffer_release'
    }

Free the resources of the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to release

Returns (Number) –

TRUE if the device could be released, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.release

    def GstAudio.AudioRingBuffer.release (self):
        #python wrapper for 'gst_audio_ring_buffer_release'

Free the resources of the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to release

Returns (bool) –

TRUE if the device could be released, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_samples_done

    guint64
    gst_audio_ring_buffer_samples_done (GstAudioRingBuffer * buf)

Get the number of samples that were processed by the ringbuffer since it was last started. This does not include the number of samples not yet processed (see gst_audio_ring_buffer_delay).

Parameters:

`buf` –

the GstAudioRingBuffer to query

Returns –

The number of samples processed by the ringbuffer.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.samples_done

    function GstAudio.AudioRingBuffer.prototype.samples_done(): {
        // javascript wrapper for 'gst_audio_ring_buffer_samples_done'
    }

Get the number of samples that were processed by the ringbuffer since it was last started. This does not include the number of samples not yet processed (see GstAudio.AudioRingBuffer.prototype.delay).

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to query

Returns (Number) –

The number of samples processed by the ringbuffer.

MT safe.

#### GstAudio.AudioRingBuffer.samples_done

    def GstAudio.AudioRingBuffer.samples_done (self):
        #python wrapper for 'gst_audio_ring_buffer_samples_done'

Get the number of samples that were processed by the ringbuffer since it was last started. This does not include the number of samples not yet processed (see GstAudio.AudioRingBuffer.delay).

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to query

Returns (int) –

The number of samples processed by the ringbuffer.

MT safe.

#### gst_audio_ring_buffer_set_callback

    gst_audio_ring_buffer_set_callback (GstAudioRingBuffer * buf,
                                        GstAudioRingBufferCallback cb,
                                        gpointer user_data)

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to set the callback on

`cb` ( [allow-none]) –

the callback to set

`user_data` –

user data passed to the callback

#### gst_audio_ring_buffer_set_callback_full

    gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer * buf,
                                             GstAudioRingBufferCallback cb,
                                             gpointer user_data,
                                             GDestroyNotify notify)

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to set the callback on

`cb` ( [allow-none]) –

the callback to set

`user_data` –

user data passed to the callback

`notify` –

function to be called when user_data is no longer needed

Since : 1.12

#### GstAudio.AudioRingBuffer.prototype.set_callback_full

    function GstAudio.AudioRingBuffer.prototype.set_callback_full(cb: GstAudio.AudioRingBufferCallback, user_data: Object): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_callback_full'
    }

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to set the callback on

`cb` (GstAudio.AudioRingBufferCallback) –

the callback to set

`user_data` (Object) –

user data passed to the callback

Since : 1.12

#### GstAudio.AudioRingBuffer.set_callback_full

    def GstAudio.AudioRingBuffer.set_callback_full (self, cb, *user_data):
        #python wrapper for 'gst_audio_ring_buffer_set_callback_full'

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to set the callback on

`cb` (GstAudio.AudioRingBufferCallback) –

the callback to set

`user_data` (variadic) –

user data passed to the callback

Since : 1.12

#### gst_audio_ring_buffer_set_channel_positions

    gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer * buf,
                                                 const GstAudioChannelPosition * position)

Tell the ringbuffer about the device's channel positions. This must be called in when the ringbuffer is acquired.

Parameters:

`buf` –

the GstAudioRingBuffer

`position` ( [array]) –

the device channel positions

#### GstAudio.AudioRingBuffer.prototype.set_channel_positions

    function GstAudio.AudioRingBuffer.prototype.set_channel_positions(position: [ GstAudio.AudioChannelPosition ]): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_channel_positions'
    }

Tell the ringbuffer about the device's channel positions. This must be called in when the ringbuffer is acquired.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`position` ([ GstAudio.AudioChannelPosition ]) –

the device channel positions

#### GstAudio.AudioRingBuffer.set_channel_positions

    def GstAudio.AudioRingBuffer.set_channel_positions (self, position):
        #python wrapper for 'gst_audio_ring_buffer_set_channel_positions'

Tell the ringbuffer about the device's channel positions. This must be called in when the ringbuffer is acquired.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer

`position` ([ GstAudio.AudioChannelPosition ]) –

the device channel positions

#### gst_audio_ring_buffer_set_errored

    gst_audio_ring_buffer_set_errored (GstAudioRingBuffer * buf)

Mark the ringbuffer as errored after it has started.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer that has encountered an error

Since : 1.24

#### GstAudio.AudioRingBuffer.prototype.set_errored

    function GstAudio.AudioRingBuffer.prototype.set_errored(): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_errored'
    }

Mark the ringbuffer as errored after it has started.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer that has encountered an error

Since : 1.24

#### GstAudio.AudioRingBuffer.set_errored

    def GstAudio.AudioRingBuffer.set_errored (self):
        #python wrapper for 'gst_audio_ring_buffer_set_errored'

Mark the ringbuffer as errored after it has started.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer that has encountered an error

Since : 1.24

#### gst_audio_ring_buffer_set_flushing

    gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer * buf,
                                        gboolean flushing)

Set the ringbuffer to flushing mode or normal mode.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to flush

`flushing` –

the new mode

#### GstAudio.AudioRingBuffer.prototype.set_flushing

    function GstAudio.AudioRingBuffer.prototype.set_flushing(flushing: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_flushing'
    }

Set the ringbuffer to flushing mode or normal mode.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to flush

`flushing` (Number) –

the new mode

#### GstAudio.AudioRingBuffer.set_flushing

    def GstAudio.AudioRingBuffer.set_flushing (self, flushing):
        #python wrapper for 'gst_audio_ring_buffer_set_flushing'

Set the ringbuffer to flushing mode or normal mode.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to flush

`flushing` (bool) –

the new mode

#### gst_audio_ring_buffer_set_sample

    gst_audio_ring_buffer_set_sample (GstAudioRingBuffer * buf,
                                      guint64 sample)

Make sure that the next sample written to the device is accounted for as being the sample sample written to the device. This value will be used in reporting the current sample position of the ringbuffer.

This function will also clear the buffer with silence.

MT safe.

Parameters:

`buf` –

the GstAudioRingBuffer to use

`sample` –

the sample number to set

#### GstAudio.AudioRingBuffer.prototype.set_sample

    function GstAudio.AudioRingBuffer.prototype.set_sample(sample: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_sample'
    }

Make sure that the next sample written to the device is accounted for as being the sample sample written to the device. This value will be used in reporting the current sample position of the ringbuffer.

This function will also clear the buffer with silence.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to use

`sample` (Number) –

the sample number to set

#### GstAudio.AudioRingBuffer.set_sample

    def GstAudio.AudioRingBuffer.set_sample (self, sample):
        #python wrapper for 'gst_audio_ring_buffer_set_sample'

Make sure that the next sample written to the device is accounted for as being the sample sample written to the device. This value will be used in reporting the current sample position of the ringbuffer.

This function will also clear the buffer with silence.

MT safe.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to use

`sample` (int) –

the sample number to set

#### gst_audio_ring_buffer_set_timestamp

    gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf,
                                         gint readseg,
                                         GstClockTime timestamp)

Parameters:

`buf` –
No description available
`readseg` –
No description available
`timestamp` –
No description available

#### GstAudio.AudioRingBuffer.prototype.set_timestamp

    function GstAudio.AudioRingBuffer.prototype.set_timestamp(readseg: Number, timestamp: Number): {
        // javascript wrapper for 'gst_audio_ring_buffer_set_timestamp'
    }

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`readseg` (Number) –
No description available
`timestamp` (Number) –
No description available

#### GstAudio.AudioRingBuffer.set_timestamp

    def GstAudio.AudioRingBuffer.set_timestamp (self, readseg, timestamp):
        #python wrapper for 'gst_audio_ring_buffer_set_timestamp'

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`readseg` (int) –
No description available
`timestamp` (int) –
No description available

#### gst_audio_ring_buffer_start

    gboolean
    gst_audio_ring_buffer_start (GstAudioRingBuffer * buf)

Start processing samples from the ringbuffer.

Parameters:

`buf` –

the GstAudioRingBuffer to start

Returns –

TRUE if the device could be started, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.start

    function GstAudio.AudioRingBuffer.prototype.start(): {
        // javascript wrapper for 'gst_audio_ring_buffer_start'
    }

Start processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to start

Returns (Number) –

TRUE if the device could be started, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.start

    def GstAudio.AudioRingBuffer.start (self):
        #python wrapper for 'gst_audio_ring_buffer_start'

Start processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to start

Returns (bool) –

TRUE if the device could be started, FALSE on error.

MT safe.

#### gst_audio_ring_buffer_stop

    gboolean
    gst_audio_ring_buffer_stop (GstAudioRingBuffer * buf)

Stop processing samples from the ringbuffer.

Parameters:

`buf` –

the GstAudioRingBuffer to stop

Returns –

TRUE if the device could be stopped, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.prototype.stop

    function GstAudio.AudioRingBuffer.prototype.stop(): {
        // javascript wrapper for 'gst_audio_ring_buffer_stop'
    }

Stop processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to stop

Returns (Number) –

TRUE if the device could be stopped, FALSE on error.

MT safe.

#### GstAudio.AudioRingBuffer.stop

    def GstAudio.AudioRingBuffer.stop (self):
        #python wrapper for 'gst_audio_ring_buffer_stop'

Stop processing samples from the ringbuffer.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –

the GstAudio.AudioRingBuffer to stop

Returns (bool) –

TRUE if the device could be stopped, FALSE on error.

MT safe.

### Functions

#### gst_audio_ring_buffer_debug_spec_buff

    gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec * spec)

Print debug info about the buffer sized in spec to the debug log.

Parameters:

`spec` –

the spec to debug

#### GstAudio.AudioRingBuffer.prototype.debug_spec_buff

    function GstAudio.AudioRingBuffer.prototype.debug_spec_buff(spec: GstAudio.AudioRingBufferSpec): {
        // javascript wrapper for 'gst_audio_ring_buffer_debug_spec_buff'
    }

Print debug info about the buffer sized in spec to the debug log.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

the spec to debug

#### GstAudio.AudioRingBuffer.debug_spec_buff

    def GstAudio.AudioRingBuffer.debug_spec_buff (spec):
        #python wrapper for 'gst_audio_ring_buffer_debug_spec_buff'

Print debug info about the buffer sized in spec to the debug log.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

the spec to debug

#### gst_audio_ring_buffer_debug_spec_caps

    gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec * spec)

Print debug info about the parsed caps in spec to the debug log.

Parameters:

`spec` –

the spec to debug

#### GstAudio.AudioRingBuffer.prototype.debug_spec_caps

    function GstAudio.AudioRingBuffer.prototype.debug_spec_caps(spec: GstAudio.AudioRingBufferSpec): {
        // javascript wrapper for 'gst_audio_ring_buffer_debug_spec_caps'
    }

Print debug info about the parsed caps in spec to the debug log.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

the spec to debug

#### GstAudio.AudioRingBuffer.debug_spec_caps

    def GstAudio.AudioRingBuffer.debug_spec_caps (spec):
        #python wrapper for 'gst_audio_ring_buffer_debug_spec_caps'

Print debug info about the parsed caps in spec to the debug log.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

the spec to debug

#### gst_audio_ring_buffer_parse_caps

    gboolean
    gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec * spec,
                                      GstCaps * caps)

Parse caps into spec.

Parameters:

`spec` –

a spec

`caps` –

a GstCaps

Returns –

TRUE if the caps could be parsed.

#### GstAudio.AudioRingBuffer.prototype.parse_caps

    function GstAudio.AudioRingBuffer.prototype.parse_caps(spec: GstAudio.AudioRingBufferSpec, caps: Gst.Caps): {
        // javascript wrapper for 'gst_audio_ring_buffer_parse_caps'
    }

Parse caps into spec.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

a spec

`caps` (Gst.Caps) –

a Gst.Caps

Returns (Number) –

TRUE if the caps could be parsed.

#### GstAudio.AudioRingBuffer.parse_caps

    def GstAudio.AudioRingBuffer.parse_caps (spec, caps):
        #python wrapper for 'gst_audio_ring_buffer_parse_caps'

Parse caps into spec.

Parameters:

`spec` (GstAudio.AudioRingBufferSpec) –

a spec

`caps` (Gst.Caps) –

a Gst.Caps

Returns (bool) –

TRUE if the caps could be parsed.

### Virtual Methods

#### acquire

    gboolean
    acquire (GstAudioRingBuffer * buf,
             GstAudioRingBufferSpec * spec)

allocate the resources for the ringbuffer using the given spec

Parameters:

`buf` –
No description available
`spec` –
No description available
Returns –
No description available

#### vfunc_acquire

    function vfunc_acquire(buf: GstAudio.AudioRingBuffer, spec: GstAudio.AudioRingBufferSpec): {
        // javascript implementation of the 'acquire' virtual method
    }

allocate the resources for the ringbuffer using the given spec

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`spec` (GstAudio.AudioRingBufferSpec) –
No description available
Returns (Number) –
No description available

#### do_acquire

    def do_acquire (buf, spec):
        #python implementation of the 'acquire' virtual method

allocate the resources for the ringbuffer using the given spec

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`spec` (GstAudio.AudioRingBufferSpec) –
No description available
Returns (bool) –
No description available

#### activate

    gboolean
    activate (GstAudioRingBuffer * buf,
              gboolean active)

activate the thread that starts pulling and monitoring the consumed segments in the device.

Parameters:

`buf` –
No description available
`active` –
No description available
Returns –
No description available

#### vfunc_activate

    function vfunc_activate(buf: GstAudio.AudioRingBuffer, active: Number): {
        // javascript implementation of the 'activate' virtual method
    }

activate the thread that starts pulling and monitoring the consumed segments in the device.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`active` (Number) –
No description available
Returns (Number) –
No description available

#### do_activate

    def do_activate (buf, active):
        #python implementation of the 'activate' virtual method

activate the thread that starts pulling and monitoring the consumed segments in the device.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`active` (bool) –
No description available
Returns (bool) –
No description available

#### clear_all

    clear_all (GstAudioRingBuffer * buf)

Optional. Clear the entire ringbuffer. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

`buf` –
No description available

#### vfunc_clear_all

    function vfunc_clear_all(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'clear_all' virtual method
    }

Optional. Clear the entire ringbuffer. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available

#### do_clear_all

    def do_clear_all (buf):
        #python implementation of the 'clear_all' virtual method

Optional. Clear the entire ringbuffer. Subclasses should chain up to the parent implementation to invoke the default handler.

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available

#### close_device

    gboolean
    close_device (GstAudioRingBuffer * buf)

close the device

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_close_device

    function vfunc_close_device(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'close_device' virtual method
    }

close the device

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_close_device

    def do_close_device (buf):
        #python implementation of the 'close_device' virtual method

close the device

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### commit

    guint
    commit (GstAudioRingBuffer * buf,
            guint64* sample,
            guint8* data,
            gint in_samples,
            gint out_samples,
            gint* accum)

write samples into the ringbuffer

Parameters:

`buf` –
No description available
`sample` –
No description available
`data` –
No description available
`in_samples` –
No description available
`out_samples` –
No description available
`accum` –
No description available
Returns –
No description available

#### vfunc_commit

    function vfunc_commit(buf: GstAudio.AudioRingBuffer, sample: Number, data: [ Number ], in_samples: Number, out_samples: Number, accum: Number): {
        // javascript implementation of the 'commit' virtual method
    }

write samples into the ringbuffer

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`sample` (Number) –
No description available
`data` ([ Number ]) –
No description available
`in_samples` (Number) –
No description available
`out_samples` (Number) –
No description available
`accum` (Number) –
No description available

Returns a tuple made of:

(Number ) –
No description available
`sample` (Number ) –
No description available
`accum` (Number ) –
No description available

#### do_commit

    def do_commit (buf, sample, data, in_samples, out_samples, accum):
        #python implementation of the 'commit' virtual method

write samples into the ringbuffer

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
`sample` (int) –
No description available
`data` ([ int ]) –
No description available
`in_samples` (int) –
No description available
`out_samples` (int) –
No description available
`accum` (int) –
No description available

Returns a tuple made of:

(int ) –
No description available
`sample` (int ) –
No description available
`accum` (int ) –
No description available

#### delay

    guint
    delay (GstAudioRingBuffer * buf)

get number of frames queued in device

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_delay

    function vfunc_delay(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'delay' virtual method
    }

get number of frames queued in device

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_delay

    def do_delay (buf):
        #python implementation of the 'delay' virtual method

get number of frames queued in device

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (int) –
No description available

#### open_device

    gboolean
    open_device (GstAudioRingBuffer * buf)

open the device, don't set any params or allocate anything

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_open_device

    function vfunc_open_device(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'open_device' virtual method
    }

open the device, don't set any params or allocate anything

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_open_device

    def do_open_device (buf):
        #python implementation of the 'open_device' virtual method

open the device, don't set any params or allocate anything

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### pause

    gboolean
    pause (GstAudioRingBuffer * buf)

pause processing of samples

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_pause

    function vfunc_pause(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'pause' virtual method
    }

pause processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_pause

    def do_pause (buf):
        #python implementation of the 'pause' virtual method

pause processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### release

    gboolean
    release (GstAudioRingBuffer * buf)

free resources of the ringbuffer

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_release

    function vfunc_release(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'release' virtual method
    }

free resources of the ringbuffer

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_release

    def do_release (buf):
        #python implementation of the 'release' virtual method

free resources of the ringbuffer

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### resume

    gboolean
    resume (GstAudioRingBuffer * buf)

resume processing of samples after pause

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_resume

    function vfunc_resume(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'resume' virtual method
    }

resume processing of samples after pause

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_resume

    def do_resume (buf):
        #python implementation of the 'resume' virtual method

resume processing of samples after pause

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### start

    gboolean
    start (GstAudioRingBuffer * buf)

start processing of samples

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_start

    function vfunc_start(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'start' virtual method
    }

start processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_start

    def do_start (buf):
        #python implementation of the 'start' virtual method

start processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

#### stop

    gboolean
    stop (GstAudioRingBuffer * buf)

stop processing of samples

Parameters:

`buf` –
No description available
Returns –
No description available

#### vfunc_stop

    function vfunc_stop(buf: GstAudio.AudioRingBuffer): {
        // javascript implementation of the 'stop' virtual method
    }

stop processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (Number) –
No description available

#### do_stop

    def do_stop (buf):
        #python implementation of the 'stop' virtual method

stop processing of samples

Parameters:

`buf` (GstAudio.AudioRingBuffer) –
No description available
Returns (bool) –
No description available

## GstAudioRingBufferSpec

The structure containing the format specification of the ringbuffer.

When type is GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD, the dsd_format is valid (otherwise it is unused). Also, when DSD is the sample type, only the rate, channels, position, and bpf fields in info are populated.

### Members

`caps` (GstCaps *) –

The caps that generated the Spec.

`type` (GstAudioRingBufferFormatType) –

the sample type

`info` (GstAudioInfo) –

the GstAudioInfo

`latency_time` (guint64) –

the latency in microseconds

`buffer_time` (guint64) –

the total buffer size in microseconds

`segsize` (gint) –

the size of one segment in bytes

`segtotal` (gint) –

the total number of segments

`seglatency` (gint) –

number of segments queued in the lower level device, defaults to segtotal

`ABI.abi.dsd_format` (GstDsdFormat) –

the GstDsdFormat (Since: 1.24)

## GstAudio.AudioRingBufferSpec

The structure containing the format specification of the ringbuffer.

When type is GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD, the dsd_format is valid (otherwise it is unused). Also, when DSD is the sample type, only the rate, channels, position, and bpf fields in info are populated.

### Members

`caps` (Gst.Caps) –

The caps that generated the Spec.

`type` (GstAudio.AudioRingBufferFormatType) –

the sample type

`info` (GstAudio.AudioInfo) –

the GstAudio.AudioInfo

`latency_time` (Number) –

the latency in microseconds

`buffer_time` (Number) –

the total buffer size in microseconds

`segsize` (Number) –

the size of one segment in bytes

`segtotal` (Number) –

the total number of segments

`seglatency` (Number) –

number of segments queued in the lower level device, defaults to segtotal

## GstAudio.AudioRingBufferSpec

The structure containing the format specification of the ringbuffer.

When type is GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD, the dsd_format is valid (otherwise it is unused). Also, when DSD is the sample type, only the rate, channels, position, and bpf fields in info are populated.

### Members

`caps` (Gst.Caps) –

The caps that generated the Spec.

`type` (GstAudio.AudioRingBufferFormatType) –

the sample type

`info` (GstAudio.AudioInfo) –

the GstAudio.AudioInfo

`latency_time` (int) –

the latency in microseconds

`buffer_time` (int) –

the total buffer size in microseconds

`segsize` (int) –

the size of one segment in bytes

`segtotal` (int) –

the total number of segments

`seglatency` (int) –

number of segments queued in the lower level device, defaults to segtotal

## Function Macros

### GST_AUDIO_RING_BUFFER_BROADCAST

    #define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf)))

### GST_AUDIO_RING_BUFFER_CAST

    #define GST_AUDIO_RING_BUFFER_CAST(obj)        ((GstAudioRingBuffer *)obj)

### GST_AUDIO_RING_BUFFER_GET_COND

    #define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond))

### GST_AUDIO_RING_BUFFER_SIGNAL

    #define GST_AUDIO_RING_BUFFER_SIGNAL(buf)   (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf)))

### GST_AUDIO_RING_BUFFER_SPEC_BUFFER_TIME

    #define GST_AUDIO_RING_BUFFER_SPEC_BUFFER_TIME(spec)   ((spec)->buffer_time)

### GST_AUDIO_RING_BUFFER_SPEC_DSD_FORMAT

    #define GST_AUDIO_RING_BUFFER_SPEC_DSD_FORMAT(spec)    ((spec)->ABI.abi.dsd_format)

### GST_AUDIO_RING_BUFFER_SPEC_FORMAT_TYPE

    #define GST_AUDIO_RING_BUFFER_SPEC_FORMAT_TYPE(spec)   ((spec)->type)

### GST_AUDIO_RING_BUFFER_SPEC_INFO

    #define GST_AUDIO_RING_BUFFER_SPEC_INFO(spec)          ((spec)->info)

### GST_AUDIO_RING_BUFFER_SPEC_LATENCY_TIME

    #define GST_AUDIO_RING_BUFFER_SPEC_LATENCY_TIME(spec)  ((spec)->latency_time)

### GST_AUDIO_RING_BUFFER_SPEC_SEGLATENCY

    #define GST_AUDIO_RING_BUFFER_SPEC_SEGLATENCY(spec)    ((spec)->seglatency)

### GST_AUDIO_RING_BUFFER_SPEC_SEGSIZE

    #define GST_AUDIO_RING_BUFFER_SPEC_SEGSIZE(spec)       ((spec)->segsize)

### GST_AUDIO_RING_BUFFER_SPEC_SEGTOTAL

    #define GST_AUDIO_RING_BUFFER_SPEC_SEGTOTAL(spec)      ((spec)->segtotal)

### GST_AUDIO_RING_BUFFER_WAIT

    #define GST_AUDIO_RING_BUFFER_WAIT(buf)     (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))

## Enumerations

### GstAudioRingBufferFormatType

The format of the samples in the ringbuffer.

##### Members

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW` (0) –

samples in linear or float

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW` (1) –

samples in mulaw

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW` (2) –

samples in alaw

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM` (3) –

samples in ima adpcm

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG` (4) –

samples in mpeg audio (but not AAC) format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM` (5) –

samples in gsm format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958` (6) –

samples in IEC958 frames (e.g. AC3)

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3` (7) –

samples in AC3 format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3` (8) –

samples in EAC3 format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS` (9) –

samples in DTS format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC` (10) –

samples in MPEG-2 AAC ADTS format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC` (11) –

samples in MPEG-4 AAC ADTS format

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW` (12) –

samples in MPEG-2 AAC raw format (Since: 1.12)

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW` (13) –

samples in MPEG-4 AAC raw format (Since: 1.12)

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC` (14) –

samples in FLAC format (Since: 1.12)

`GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD` (15) –

samples in DSD format (Since: 1.24)

### GstAudio.AudioRingBufferFormatType

The format of the samples in the ringbuffer.

##### Members

`GstAudio.AudioRingBufferFormatType.RAW` (0) –

samples in linear or float

`GstAudio.AudioRingBufferFormatType.MU_LAW` (1) –

samples in mulaw

`GstAudio.AudioRingBufferFormatType.A_LAW` (2) –

samples in alaw

`GstAudio.AudioRingBufferFormatType.IMA_ADPCM` (3) –

samples in ima adpcm

`GstAudio.AudioRingBufferFormatType.MPEG` (4) –

samples in mpeg audio (but not AAC) format

`GstAudio.AudioRingBufferFormatType.GSM` (5) –

samples in gsm format

`GstAudio.AudioRingBufferFormatType.IEC958` (6) –

samples in IEC958 frames (e.g. AC3)

`GstAudio.AudioRingBufferFormatType.AC3` (7) –

samples in AC3 format

`GstAudio.AudioRingBufferFormatType.EAC3` (8) –

samples in EAC3 format

`GstAudio.AudioRingBufferFormatType.DTS` (9) –

samples in DTS format

`GstAudio.AudioRingBufferFormatType.MPEG2_AAC` (10) –

samples in MPEG-2 AAC ADTS format

`GstAudio.AudioRingBufferFormatType.MPEG4_AAC` (11) –

samples in MPEG-4 AAC ADTS format

`GstAudio.AudioRingBufferFormatType.MPEG2_AAC_RAW` (12) –

samples in MPEG-2 AAC raw format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.MPEG4_AAC_RAW` (13) –

samples in MPEG-4 AAC raw format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.FLAC` (14) –

samples in FLAC format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.DSD` (15) –

samples in DSD format (Since: 1.24)

### GstAudio.AudioRingBufferFormatType

The format of the samples in the ringbuffer.

##### Members

`GstAudio.AudioRingBufferFormatType.RAW` (0) –

samples in linear or float

`GstAudio.AudioRingBufferFormatType.MU_LAW` (1) –

samples in mulaw

`GstAudio.AudioRingBufferFormatType.A_LAW` (2) –

samples in alaw

`GstAudio.AudioRingBufferFormatType.IMA_ADPCM` (3) –

samples in ima adpcm

`GstAudio.AudioRingBufferFormatType.MPEG` (4) –

samples in mpeg audio (but not AAC) format

`GstAudio.AudioRingBufferFormatType.GSM` (5) –

samples in gsm format

`GstAudio.AudioRingBufferFormatType.IEC958` (6) –

samples in IEC958 frames (e.g. AC3)

`GstAudio.AudioRingBufferFormatType.AC3` (7) –

samples in AC3 format

`GstAudio.AudioRingBufferFormatType.EAC3` (8) –

samples in EAC3 format

`GstAudio.AudioRingBufferFormatType.DTS` (9) –

samples in DTS format

`GstAudio.AudioRingBufferFormatType.MPEG2_AAC` (10) –

samples in MPEG-2 AAC ADTS format

`GstAudio.AudioRingBufferFormatType.MPEG4_AAC` (11) –

samples in MPEG-4 AAC ADTS format

`GstAudio.AudioRingBufferFormatType.MPEG2_AAC_RAW` (12) –

samples in MPEG-2 AAC raw format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.MPEG4_AAC_RAW` (13) –

samples in MPEG-4 AAC raw format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.FLAC` (14) –

samples in FLAC format (Since: 1.12)

`GstAudio.AudioRingBufferFormatType.DSD` (15) –

samples in DSD format (Since: 1.24)

### GstAudioRingBufferState

The state of the ringbuffer.

##### Members

`GST_AUDIO_RING_BUFFER_STATE_STOPPED` (0) –

The ringbuffer is stopped

`GST_AUDIO_RING_BUFFER_STATE_PAUSED` (1) –

The ringbuffer is paused

`GST_AUDIO_RING_BUFFER_STATE_STARTED` (2) –

The ringbuffer is started

`GST_AUDIO_RING_BUFFER_STATE_ERROR` (3) –

The ringbuffer has encountered an error after it has been started, e.g. because the device was disconnected (Since: 1.2)

### GstAudio.AudioRingBufferState

The state of the ringbuffer.

##### Members

`GstAudio.AudioRingBufferState.STOPPED` (0) –

The ringbuffer is stopped

`GstAudio.AudioRingBufferState.PAUSED` (1) –

The ringbuffer is paused

`GstAudio.AudioRingBufferState.STARTED` (2) –

The ringbuffer is started

`GstAudio.AudioRingBufferState.ERROR` (3) –

The ringbuffer has encountered an error after it has been started, e.g. because the device was disconnected (Since: 1.2)

### GstAudio.AudioRingBufferState

The state of the ringbuffer.

##### Members

`GstAudio.AudioRingBufferState.STOPPED` (0) –

The ringbuffer is stopped

`GstAudio.AudioRingBufferState.PAUSED` (1) –

The ringbuffer is paused

`GstAudio.AudioRingBufferState.STARTED` (2) –

The ringbuffer is started

`GstAudio.AudioRingBufferState.ERROR` (3) –

The ringbuffer has encountered an error after it has been started, e.g. because the device was disconnected (Since: 1.2)

## Callbacks

### GstAudioRingBufferCallback

    (*GstAudioRingBufferCallback) (GstAudioRingBuffer * rbuf,
                                   guint8* data,
                                   guint len,
                                   gpointer user_data)

This function is set with gst_audio_ring_buffer_set_callback and is called to fill the memory at data with len bytes of samples.

Parameters:

`rbuf` –

a GstAudioRingBuffer

`data` ( [arraylength=len]) –

target to fill

`len` –

amount to fill

`user_data` –

user data

### GstAudio.AudioRingBufferCallback

    function GstAudio.AudioRingBufferCallback(rbuf: GstAudio.AudioRingBuffer, data: [ Number ], len: Number, user_data: Object): {
        // javascript wrapper for 'GstAudioRingBufferCallback'
    }

This function is set with gst_audio_ring_buffer_set_callback (not introspectable) and is called to fill the memory at data with len bytes of samples.

Parameters:

`rbuf` (GstAudio.AudioRingBuffer) –

a GstAudio.AudioRingBuffer

`data` ([ Number ]) –

target to fill

`len` (Number) –

amount to fill

`user_data` (Object) –

user data

### GstAudio.AudioRingBufferCallback

    def GstAudio.AudioRingBufferCallback (rbuf, data, len, *user_data):
        #python wrapper for 'GstAudioRingBufferCallback'

This function is set with gst_audio_ring_buffer_set_callback (not introspectable) and is called to fill the memory at data with len bytes of samples.

Parameters:

`rbuf` (GstAudio.AudioRingBuffer) –

a GstAudio.AudioRingBuffer

`data` ([ int ]) –

target to fill

`len` (int) –

amount to fill

`user_data` (variadic) –

user data

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