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gint64 | actual-buffer-time | Read |
gint64 | actual-latency-time | Read |
gint64 | buffer-time | Read / Write |
gint64 | latency-time | Read / Write |
gboolean | provide-clock | Read / Write |
GstAudioBaseSrcSlaveMethod | slave-method | Read / Write |
GObject ╰── GInitiallyUnowned ╰── GstObject ╰── GstElement ╰── GstBaseSrc ╰── GstPushSrc ╰── GstAudioBaseSrc ╰── GstAudioSrc
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.
#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
Get the GstClock of obj
.
#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
Get the source GstPad of obj
.
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()
).
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.
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.
GstAudioBaseSrcSlaveMethod
gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src
);
Get 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
.
struct GstAudioBaseSrcClass { GstPushSrcClass parent_class; /* subclass ringbuffer allocation */ GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src); };
GstAudioBaseSrc class. Override the vmethod to implement functionality.
“actual-buffer-time”
property“actual-buffer-time” gint64
Actual configured size of audio buffer in microseconds.
Flags: Read
Allowed values: >= -1
Default value: -1
“actual-latency-time”
property“actual-latency-time” gint64
Actual configured audio latency in microseconds.
Flags: Read
Allowed values: >= -1
Default value: -1
“buffer-time”
property“buffer-time” gint64
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.
Flags: Read / Write
Allowed values: >= 1
Default value: 200000
“latency-time”
property“latency-time” gint64
The minimum amount of data to read in each iteration in microseconds. This is the minimum latency that the source reports.
Flags: Read / Write
Allowed values: >= 1
Default value: 10000
“provide-clock”
property“provide-clock” gboolean
Provide a clock to be used as the global pipeline clock.
Flags: Read / Write
Default value: TRUE
“slave-method”
property“slave-method” GstAudioBaseSrcSlaveMethod
Algorithm used to match the rate of the masterclock.
Flags: Read / Write
Default value: GST_AUDIO_BASE_SRC_SLAVE_SKEW