|  |  |  | GStreamer Base Plugins 0.10 Library Reference Manual |  | 
|---|---|---|---|---|
#include <gst/audio/gstbaseaudiosrc.h>
                    GstBaseAudioSrc;
                    GstBaseAudioSrcClass;
#define             GST_BASE_AUDIO_SRC_CLOCK            (obj)
#define             GST_BASE_AUDIO_SRC_PAD              (obj)
GstRingBuffer*      gst_base_audio_src_create_ringbuffer
                                                        (GstBaseAudioSrc *src);
void                gst_base_audio_src_set_provide_clock
                                                        (GstBaseAudioSrc *src,
                                                         gboolean provide);
gboolean            gst_base_audio_src_get_provide_clock
                                                        (GstBaseAudioSrc *src);
  GObject
   +----GstObject
         +----GstElement
               +----GstBaseSrc
                     +----GstPushSrc
                           +----GstBaseAudioSrc
                                 +----GstAudioSrc
"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" GstBaseAudioSrcSlaveMethod : Read / Write
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)
typedef struct {
  GstPushSrcClass parent_class;
  /* subclass ringbuffer allocation */
  GstRingBuffer* (*create_ringbuffer)  (GstBaseAudioSrc *src);
} GstBaseAudioSrcClass;
GstBaseAudioSrc class. Override the vmethod to implement functionality.
| GstPushSrcClass  | the parent class. | 
| 
 | create and return a GstRingBuffer to read from. | 
#define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock)
Get the GstClock of obj.
| 
 | a GstBaseAudioSrc | 
#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
Get the source GstPad of obj.
| 
 | a GstBaseAudioSrc | 
GstRingBuffer* gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
Create and return the GstRingBuffer 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()).
| 
 | a GstBaseAudioSrc. | 
| Returns : | The new ringbuffer of src. | 
void                gst_base_audio_src_set_provide_clock
                                                        (GstBaseAudioSrc *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.
| 
 | a GstBaseAudioSrc | 
| 
 | new state | 
Since 0.10.16
gboolean            gst_base_audio_src_get_provide_clock
                                                        (GstBaseAudioSrc *src);
Queries whether src will provide a clock or not. See also
gst_base_audio_src_set_provide_clock.
| 
 | a GstBaseAudioSrc | 
| Returns : | TRUEifsrcwill provide a clock. | 
Since 0.10.16
"actual-buffer-time" property"actual-buffer-time" gint64 : Read
Actual configured size of audio buffer in microseconds.
Allowed values: >= -1
Default value: -1
Since 0.10.20
"actual-latency-time" property"actual-latency-time" gint64 : Read
Actual configured audio latency in microseconds.
Allowed values: >= -1
Default value: -1
Since 0.10.20
"buffer-time" property"buffer-time" gint64 : Read / Write
Size of audio buffer in microseconds.
Allowed values: >= 1
Default value: 200000
"latency-time" property"latency-time" gint64 : Read / Write
Audio latency in microseconds.
Allowed values: >= 1
Default value: 10000
"provide-clock" property"provide-clock" gboolean : Read / Write
Provide a clock to be used as the global pipeline clock.
Default value: TRUE