gstaudiobasesrc

gstaudiobasesrc — Base class for audio sources

Synopsis

#include <gst/audio/gstaudiobasesrc.h>

struct              GstAudioBaseSrc;
struct              GstAudioBaseSrcClass;
enum                GstAudioBaseSrcSlaveMethod;
#define             GST_AUDIO_BASE_SRC_CLOCK            (obj)
#define             GST_AUDIO_BASE_SRC_PAD              (obj)
GstAudioRingBuffer * gst_audio_base_src_create_ringbuffer
                                                        (GstAudioBaseSrc *src);
void                gst_audio_base_src_set_provide_clock
                                                        (GstAudioBaseSrc *src,
                                                         gboolean provide);
gboolean            gst_audio_base_src_get_provide_clock
                                                        (GstAudioBaseSrc *src);
GstAudioBaseSrcSlaveMethod gst_audio_base_src_get_slave_method
                                                        (GstAudioBaseSrc *src);
void                gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
                                                         GstAudioBaseSrcSlaveMethod method);

Object Hierarchy

  GObject
   +----GInitiallyUnowned
         +----GstObject
               +----GstElement
                     +----GstBaseSrc
                           +----GstPushSrc
                                 +----GstAudioBaseSrc
                                       +----GstAudioSrc

Properties

  "actual-buffer-time"       gint64                : Read
  "actual-latency-time"      gint64                : Read
  "buffer-time"              gint64                : Read / Write
  "latency-time"             gint64                : Read / Write
  "provide-clock"            gboolean              : Read / Write
  "slave-method"             GstAudioBaseSrcSlaveMethod  : Read / Write

Description

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.

Last reviewed on 2006-09-27 (0.10.12)

Details

struct GstAudioBaseSrc

struct GstAudioBaseSrc;

Opaque GstAudioBaseSrc.


struct GstAudioBaseSrcClass

struct GstAudioBaseSrcClass {
  GstPushSrcClass      parent_class;

  /* subclass ringbuffer allocation */
  GstAudioRingBuffer* (*create_ringbuffer)  (GstAudioBaseSrc *src);
};

GstAudioBaseSrc class. Override the vmethod to implement functionality.

GstPushSrcClass parent_class;

the parent class.

create_ringbuffer ()

create and return a GstAudioRingBuffer to read from.

enum GstAudioBaseSrcSlaveMethod

typedef enum {
  GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
  GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP,
  GST_AUDIO_BASE_SRC_SLAVE_SKEW,
  GST_AUDIO_BASE_SRC_SLAVE_NONE
} GstAudioBaseSrcSlaveMethod;

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

GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE

Resample to match the master clock.

GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP

Retimestamp output buffers with master clock time.

GST_AUDIO_BASE_SRC_SLAVE_SKEW

Adjust capture pointer when master clock drifts too much.

GST_AUDIO_BASE_SRC_SLAVE_NONE

No adjustment is done.

GST_AUDIO_BASE_SRC_CLOCK()

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

Get the GstClock of obj.

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.

obj :

a GstAudioBaseSrc

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()).

src :

a GstAudioBaseSrc.

Returns :

The new ringbuffer of src. [transfer none]

gst_audio_base_src_set_provide_clock ()

void                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.

src :

a GstAudioBaseSrc

provide :

new state

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.

src :

a GstAudioBaseSrc

Returns :

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.

src :

a GstAudioBaseSrc

Returns :

The current slave method used by src.

gst_audio_base_src_set_slave_method ()

void                gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
                                                         GstAudioBaseSrcSlaveMethod method);

Controls how clock slaving will be performed in src.

src :

a GstAudioBaseSrc

method :

the new slave method

Property Details

The "actual-buffer-time" property

  "actual-buffer-time"       gint64                : Read

Actual configured size of audio buffer in microseconds.

Allowed values: >= -1

Default value: -1


The "actual-latency-time" property

  "actual-latency-time"      gint64                : Read

Actual configured audio latency in microseconds.

Allowed values: >= -1

Default value: -1


The "buffer-time" property

  "buffer-time"              gint64                : Read / Write

Size of audio buffer in microseconds, this is the maximum amount of data that is buffered in the device and the maximum latency that the source reports.

Allowed values: >= 1

Default value: 200000


The "latency-time" property

  "latency-time"             gint64                : Read / Write

The minimum amount of data to read in each iteration in microseconds, this is the minimum latency that the source reports.

Allowed values: >= 1

Default value: 10000


The "provide-clock" property

  "provide-clock"            gboolean              : Read / Write

Provide a clock to be used as the global pipeline clock.

Default value: TRUE


The "slave-method" property

  "slave-method"             GstAudioBaseSrcSlaveMethod  : Read / Write

Algorithm to use to match the rate of the masterclock.

Default value: GST_AUDIO_BASE_SRC_SLAVE_SKEW

See Also

GstAudioSrc, GstAudioRingBuffer.