GStreamer Base Plugins 0.10 Library Reference Manual | ||||
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Top | Description | Object Hierarchy | Properties |
#include <gst/audio/gstbaseaudiosrc.h> struct GstBaseAudioSrc; struct GstBaseAudioSrcClass; enum GstBaseAudioSrcSlaveMethod; #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
); GstBaseAudioSrcSlaveMethod gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src
); void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src
,GstBaseAudioSrcSlaveMethod method
);
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)
struct GstBaseAudioSrcClass { GstPushSrcClass parent_class; /* subclass ringbuffer allocation */ GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src); };
GstBaseAudioSrc class. Override the vmethod to implement functionality.
the parent class. | |
create and return a GstRingBuffer to read from. |
typedef enum { GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE, GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP, GST_BASE_AUDIO_SRC_SLAVE_SKEW, GST_BASE_AUDIO_SRC_SLAVE_NONE } GstBaseAudioSrcSlaveMethod;
Different possible clock slaving algorithms when the internal audio clock was not selected as the pipeline clock.
#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 : |
TRUE if src will provide a clock. |
Since 0.10.16
GstBaseAudioSrcSlaveMethod gst_base_audio_src_get_slave_method
(GstBaseAudioSrc *src
);
Get the current slave method used by src
.
|
a GstBaseAudioSrc |
Returns : |
The current slave method used by src . |
Since 0.10.20
void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src
,GstBaseAudioSrcSlaveMethod method
);
Controls how clock slaving will be performed in src
.
|
a GstBaseAudioSrc |
|
the new slave method |
Since 0.10.20
"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
"slave-method"
property"slave-method" GstBaseAudioSrcSlaveMethod : Read / Write
Algorithm to use to match the rate of the masterclock.
Default value: GST_BASE_AUDIO_SRC_SLAVE_SKEW