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)
		–
	ringbuffer
		(GstAudioRingBuffer *)
		–
	buffer_time
		(GstClockTime)
		–
	latency_time
		(GstClockTime)
		–
	next_sample
		(guint64)
		–
	clock
		(GstClock *)
		–
	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.
GstAudio.AudioBaseSrc
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──Gst.Element ╰──GstBase.BaseSrc ╰──GstBase.PushSrc ╰──GstAudio.AudioBaseSrc ╰──GstAudio.AudioSrc
Opaque GstAudio.AudioBaseSrc.
Members
element
		(GstBase.PushSrc)
		–
	ringbuffer
		(GstAudio.AudioRingBuffer)
		–
	buffer_time
		(Number)
		–
	latency_time
		(Number)
		–
	next_sample
		(Number)
		–
	clock
		(Gst.Clock)
		–
	GstAudio.AudioBaseSrc
GObject.Object ╰──GObject.InitiallyUnowned ╰──Gst.Object ╰──Gst.Element ╰──GstBase.BaseSrc ╰──GstBase.PushSrc ╰──GstAudio.AudioBaseSrc ╰──GstAudio.AudioSrc
Opaque GstAudio.AudioBaseSrc.
Members
element
		(GstBase.PushSrc)
		–
	ringbuffer
		(GstAudio.AudioRingBuffer)
		–
	buffer_time
		(int)
		–
	latency_time
		(int)
		–
	next_sample
		(int)
		–
	clock
		(Gst.Clock)
		–
	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
–
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:
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:
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
–
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:
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:
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
–
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.
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.
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.
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.
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.
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.
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
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
Properties
actual-buffer-time
“actual-buffer-time” gint64
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_buffer_time
“self.props.actual_buffer_time” int
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
“actual-latency-time” Number
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
Virtual Methods
create_ringbuffer
GstAudioRingBuffer * create_ringbuffer (GstAudioBaseSrc * src)
create and return a GstAudioRingBuffer to read from.
Parameters:
src
–
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:
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:
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
–
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
–
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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