GstAudioBaseSrc

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

GstAudioBaseSrc

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBaseSrc
                    ╰──GstPushSrc
                        ╰──GstAudioBaseSrc
                            ╰──GstAudioSrc

Opaque GstAudioBaseSrc.

Members

element (GstPushSrc) –
No description available
ringbuffer (GstAudioRingBuffer *) –
No description available
buffer_time (GstClockTime) –
No description available
latency_time (GstClockTime) –
No description available
next_sample (guint64) –
No description available
clock (GstClock *) –
No description available

Class structure

GstAudioBaseSrcClass

GstAudioBaseSrc class. Override the vmethod to implement functionality.

Fields
parent_class (GstPushSrcClass) –

the parent class.


GstAudio.AudioBaseSrcClass

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

Attributes
parent_class (GstBase.PushSrcClass) –

the parent class.


GstAudio.AudioBaseSrcClass

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

Attributes
parent_class (GstBase.PushSrcClass) –

the parent class.


GstAudioBaseSrc

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBaseSrc
                    ╰──GstPushSrc
                        ╰──GstAudioBaseSrc
                            ╰──GstAudioSrc

Opaque GstAudioBaseSrc.

Members

element (GstPushSrc) –
No description available
ringbuffer (GstAudioRingBuffer) –
No description available
buffer_time (GstClockTime) –
No description available
latency_time (GstClockTime) –
No description available
next_sample (int) –
No description available
clock (GstClock) –
No description available

GstAudioBaseSrc

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBaseSrc
                    ╰──GstPushSrc
                        ╰──GstAudioBaseSrc
                            ╰──GstAudioSrc

Opaque GstAudioBaseSrc.

Members

element (GstPushSrc) –
No description available
ringbuffer (GstAudioRingBuffer) –
No description available
buffer_time (GstClockTime) –
No description available
latency_time (GstClockTime) –
No description available
next_sample (Number) –
No description available
clock (GstClock) –
No description available

Methods

gst_audio_base_src_create_ringbuffer

GstAudioRingBuffer *
gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc * src)

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

Parameters:

src

a GstAudioBaseSrc.

Returns ( [transfer: none] ) –

The new ringbuffer of src.


GstAudio.AudioBaseSrc.create_ringbuffer

def GstAudio.AudioBaseSrc.create_ringbuffer (self):
    #python wrapper for 'gst_audio_base_src_create_ringbuffer'

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

Parameters:

Returns ( GstAudio.AudioRingBuffer ) –

The new ringbuffer of src.


GstAudio.AudioBaseSrc.prototype.create_ringbuffer

function GstAudio.AudioBaseSrc.prototype.create_ringbuffer(): {
    // javascript wrapper for 'gst_audio_base_src_create_ringbuffer'
}

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

Parameters:

Returns ( GstAudio.AudioRingBuffer ) –

The new ringbuffer of src.


gst_audio_base_src_get_provide_clock

gboolean
gst_audio_base_src_get_provide_clock (GstAudioBaseSrc * src)

Queries whether src will provide a clock or not. See also gst_audio_base_src_set_provide_clock.

Parameters:

src

a GstAudioBaseSrc

Returns

TRUE if src will provide a clock.


GstAudio.AudioBaseSrc.get_provide_clock

def GstAudio.AudioBaseSrc.get_provide_clock (self):
    #python wrapper for 'gst_audio_base_src_get_provide_clock'

Queries whether src will provide a clock or not. See also gst_audio_base_src_set_provide_clock.

Parameters:

Returns ( bool ) –

True if src will provide a clock.


GstAudio.AudioBaseSrc.prototype.get_provide_clock

function GstAudio.AudioBaseSrc.prototype.get_provide_clock(): {
    // javascript wrapper for 'gst_audio_base_src_get_provide_clock'
}

Queries whether src will provide a clock or not. See also gst_audio_base_src_set_provide_clock.

Parameters:

Returns ( Number ) –

true if src will provide a clock.


gst_audio_base_src_get_slave_method

GstAudioBaseSrcSlaveMethod
gst_audio_base_src_get_slave_method (GstAudioBaseSrc * src)

Get the current slave method used by src.

Parameters:

src

a GstAudioBaseSrc

Returns

The current slave method used by src.


GstAudio.AudioBaseSrc.get_slave_method

def GstAudio.AudioBaseSrc.get_slave_method (self):
    #python wrapper for 'gst_audio_base_src_get_slave_method'

Get the current slave method used by src.

Parameters:

The current slave method used by src.


GstAudio.AudioBaseSrc.prototype.get_slave_method

function GstAudio.AudioBaseSrc.prototype.get_slave_method(): {
    // javascript wrapper for 'gst_audio_base_src_get_slave_method'
}

Get the current slave method used by src.

Parameters:

The current slave method used by src.


gst_audio_base_src_set_provide_clock

gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src,
                                      gboolean provide)

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

Parameters:

src

a GstAudioBaseSrc

provide

new state


GstAudio.AudioBaseSrc.set_provide_clock

def GstAudio.AudioBaseSrc.set_provide_clock (self, provide):
    #python wrapper for 'gst_audio_base_src_set_provide_clock'

Controls whether src will provide a clock or not. If provide is True, Gst.Element.provide_clock will return a clock that reflects the datarate of src. If provide is False, Gst.Element.provide_clock will return NULL.

Parameters:

provide ( bool ) –

new state


GstAudio.AudioBaseSrc.prototype.set_provide_clock

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

Controls whether src will provide a clock or not. If provide is true, Gst.Element.prototype.provide_clock will return a clock that reflects the datarate of src. If provide is false, Gst.Element.prototype.provide_clock will return NULL.

Parameters:

provide ( Number ) –

new state


gst_audio_base_src_set_slave_method

gst_audio_base_src_set_slave_method (GstAudioBaseSrc * src,
                                     GstAudioBaseSrcSlaveMethod method)

Controls how clock slaving will be performed in src.

Parameters:

src

a GstAudioBaseSrc

method

the new slave method


GstAudio.AudioBaseSrc.set_slave_method

def GstAudio.AudioBaseSrc.set_slave_method (self, method):
    #python wrapper for 'gst_audio_base_src_set_slave_method'

Controls how clock slaving will be performed in src.

Parameters:

the new slave method


GstAudio.AudioBaseSrc.prototype.set_slave_method

function GstAudio.AudioBaseSrc.prototype.set_slave_method(method: GstAudio.AudioBaseSrcSlaveMethod): {
    // javascript wrapper for 'gst_audio_base_src_set_slave_method'
}

Controls how clock slaving will be performed in src.

Parameters:

the new slave method


Properties

actual-buffer-time

“actual-buffer-time” gint64

Actual configured size of audio buffer in microseconds.

Flags : Read


actual_buffer_time

“self.props.actual_buffer_time” int

Actual configured size of audio buffer in microseconds.

Flags : Read


actual-buffer-time

“actual-buffer-time” Number

Actual configured size of audio buffer in microseconds.

Flags : Read


actual-latency-time

“actual-latency-time” gint64

Actual configured audio latency in microseconds.

Flags : Read


actual_latency_time

“self.props.actual_latency_time” int

Actual configured audio latency in microseconds.

Flags : Read


actual-latency-time

“actual-latency-time” Number

Actual configured audio latency in microseconds.

Flags : Read


buffer-time

“buffer-time” gint64

Flags : Read / Write


buffer_time

“self.props.buffer_time” int

Flags : Read / Write


buffer-time

“buffer-time” Number

Flags : Read / Write


latency-time

“latency-time” gint64

Flags : Read / Write


latency_time

“self.props.latency_time” int

Flags : Read / Write


latency-time

“latency-time” Number

Flags : Read / Write


provide-clock

“provide-clock” gboolean

Flags : Read / Write


provide_clock

“self.props.provide_clock” bool

Flags : Read / Write


provide-clock

“provide-clock” Number

Flags : Read / Write


slave-method

“slave-method” GstAudioBaseSrcSlaveMethod *

Flags : Read / Write


slave_method

“self.props.slave_method” GstAudio.AudioBaseSrcSlaveMethod

Flags : Read / Write


slave-method

“slave-method” GstAudio.AudioBaseSrcSlaveMethod

Flags : Read / Write


Virtual Methods

create_ringbuffer

GstAudioRingBuffer *
create_ringbuffer (GstAudioBaseSrc * src)

create and return a GstAudioRingBuffer to read from.

Parameters:

src
No description available
Returns
No description available

do_create_ringbuffer

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

create and return a GstAudio.AudioRingBuffer to read from.

Parameters:

src ( GstAudio.AudioBaseSrc ) –
No description available
Returns ( GstAudio.AudioRingBuffer ) –
No description available

vfunc_create_ringbuffer

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

create and return a GstAudio.AudioRingBuffer to read from.

Parameters:

src ( GstAudio.AudioBaseSrc ) –
No description available
Returns ( GstAudio.AudioRingBuffer ) –
No description available

Function Macros

GST_AUDIO_BASE_SRC_CAST

#define GST_AUDIO_BASE_SRC_CAST(obj)            ((GstAudioBaseSrc*)obj)

GST_AUDIO_BASE_SRC_CLOCK

#define GST_AUDIO_BASE_SRC_CLOCK(obj)    (GST_AUDIO_BASE_SRC (obj)->clock)

Get the GstClock of obj.

Parameters:

obj

a GstAudioBaseSrc


GST_AUDIO_BASE_SRC_PAD

#define GST_AUDIO_BASE_SRC_PAD(obj)      (GST_BASE_SRC (obj)->srcpad)

Get the source GstPad of obj.

Parameters:

obj

a GstAudioBaseSrc


Enumerations

GstAudioBaseSrcSlaveMethod

Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.

Members

GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE (0) –

Resample to match the master clock.

GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP (1) –

Retimestamp output buffers with master clock time.

GST_AUDIO_BASE_SRC_SLAVE_SKEW (2) –

Adjust capture pointer when master clock drifts too much.

GST_AUDIO_BASE_SRC_SLAVE_NONE (3) –

No adjustment is done.


GstAudio.AudioBaseSrcSlaveMethod

Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.

Members

GstAudio.AudioBaseSrcSlaveMethod.RESAMPLE (0) –

Resample to match the master clock.

GstAudio.AudioBaseSrcSlaveMethod.RE_TIMESTAMP (1) –

Retimestamp output buffers with master clock time.

GstAudio.AudioBaseSrcSlaveMethod.SKEW (2) –

Adjust capture pointer when master clock drifts too much.

GstAudio.AudioBaseSrcSlaveMethod.NONE (3) –

No adjustment is done.


GstAudio.AudioBaseSrcSlaveMethod

Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.

Members

GstAudio.AudioBaseSrcSlaveMethod.RESAMPLE (0) –

Resample to match the master clock.

GstAudio.AudioBaseSrcSlaveMethod.RE_TIMESTAMP (1) –

Retimestamp output buffers with master clock time.

GstAudio.AudioBaseSrcSlaveMethod.SKEW (2) –

Adjust capture pointer when master clock drifts too much.

GstAudio.AudioBaseSrcSlaveMethod.NONE (3) –

No adjustment is done.


Constants

GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP

#define GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP

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