Gst.Audio.BaseSrc – gstreamer-audio-1.0

Description:

[ CCode ( type_id = "gst_audio_base_src_get_type ()" ) ]
[ GIR ( name = "AudioBaseSrc" ) ]
public class BaseSrc : PushSrc

This is the base class for audio sources.

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

All known sub-classes:

Namespace: Gst.Audio

Content:

Properties:

public int64 actual_buffer_time { get; }

Actual configured size of audio buffer in microseconds.

public int64 actual_latency_time { get; }

Actual configured audio latency in microseconds.

public int64 buffer_time { get; set; }

public int64 latency_time { get; set; }

public bool provide_clock { get; set; }

public BaseSrcSlaveMethod slave_method { get; set; }

Creation methods:

protected BaseSrc ()

Methods:

public virtual unowned RingBuffer create_ringbuffer ()

Create and return the RingBuffer for this.

public bool get_provide_clock ()

Queries whether this will provide a clock or not.

public BaseSrcSlaveMethod get_slave_method ()

Get the current slave method used by this .

public void set_provide_clock (bool provide)

Controls whether this will provide a clock or not.

public void set_slave_method (BaseSrcSlaveMethod method)

Controls how clock slaving will be performed in this.

Fields:

Inherited Members:

All known members inherited from class Gst.Base.PushSrc

alloc

create

fill

BaseSrc