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GStreamer Base Plugins 1.0 Library Reference Manual | ![]() |
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Top | Description | Object Hierarchy | Properties |
#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
);
GObject +----GInitiallyUnowned +----GstObject +----GstElement +----GstBaseSrc +----GstPushSrc +----GstAudioBaseSrc +----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" GstAudioBaseSrcSlaveMethod : 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)
struct GstAudioBaseSrcClass { GstPushSrcClass parent_class; /* subclass ringbuffer allocation */ GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src); };
GstAudioBaseSrc class. Override the vmethod to implement functionality.
the parent class. | |
create and return a GstAudioRingBuffer to read from. |
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.
#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
Get the GstClock of obj
.
|
a GstAudioBaseSrc |
#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
Get the source GstPad of obj
.
|
a GstAudioBaseSrc |
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()
).
|
a GstAudioBaseSrc. |
Returns : |
The new ringbuffer of src . [transfer none]
|
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.
|
a GstAudioBaseSrc |
|
new state |
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.
|
a GstAudioBaseSrc |
Returns : |
TRUE if src will provide a clock. |
GstAudioBaseSrcSlaveMethod gst_audio_base_src_get_slave_method
(GstAudioBaseSrc *src
);
Get the current slave method used by src
.
|
a GstAudioBaseSrc |
Returns : |
The current slave method used by src . |
void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src
,GstAudioBaseSrcSlaveMethod method
);
Controls how clock slaving will be performed in src
.
|
a GstAudioBaseSrc |
|
the new slave method |
"actual-buffer-time"
property "actual-buffer-time" gint64 : Read
Actual configured size of audio buffer in microseconds.
Allowed values: >= -1
Default value: -1
"actual-latency-time"
property "actual-latency-time" gint64 : Read
Actual configured audio latency in microseconds.
Allowed values: >= -1
Default value: -1
"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
"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
"provide-clock"
property "provide-clock" gboolean : Read / Write
Provide a clock to be used as the global pipeline clock.
Default value: TRUE
"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