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 parent_class ; |
the parent class. |
create_ringbuffer () |
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
.
obj : |
a GstBaseAudioSrc |
#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
Get the source GstPad of obj
.
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()
).
src : |
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.
src : |
a GstBaseAudioSrc |
provide : |
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.
src : |
a GstBaseAudioSrc |
Returns : | TRUE if src will 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: 40000
"provide-clock"
property"provide-clock" gboolean : Read / Write
Provide a clock to be used as the global pipeline clock.
Default value: TRUE