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<title>GstAudioBaseSrc: GStreamer Base Plugins 1.0 Library Reference Manual</title>
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Top  | 
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                  Description  | 
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                  Object Hierarchy  | 
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                  Properties
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GstAudioBaseSrc
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GstAudioBaseSrc — Base class for audio sources
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Functions
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#define
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GST_AUDIO_BASE_SRC_CLOCK()
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#define
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GST_AUDIO_BASE_SRC_PAD()
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GstAudioRingBuffer *
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gst_audio_base_src_create_ringbuffer ()
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void
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gst_audio_base_src_set_provide_clock ()
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gboolean
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gst_audio_base_src_get_provide_clock ()
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GstAudioBaseSrcSlaveMethod
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gst_audio_base_src_get_slave_method ()
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void
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gst_audio_base_src_set_slave_method ()
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Properties
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gint64
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actual-buffer-time
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Read
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gint64
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actual-latency-time
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Read
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gint64
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buffer-time
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Read / Write
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gint64
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latency-time
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Read / Write
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gboolean
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provide-clock
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Read / Write
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GstAudioBaseSrcSlaveMethod
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slave-method
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Read / Write
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Types and Values
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struct
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GstAudioBaseSrc
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struct
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GstAudioBaseSrcClass
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enum
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GstAudioBaseSrcSlaveMethod
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Object Hierarchy
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    GEnum
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    ╰── GstAudioBaseSrcSlaveMethod
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    GObject
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    ╰── GInitiallyUnowned
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        ╰── GstObject
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            ╰── GstElement
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                ╰── GstBaseSrc
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                    ╰── GstPushSrc
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                        ╰── GstAudioBaseSrc
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                            ╰── GstAudioSrc
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Includes
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#include <gst/audio/audio.h>
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Description
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This is the base class for audio sources. Subclasses need to implement the
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::create_ringbuffer vmethod. This base class will then take care of
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reading samples from the ringbuffer, synchronisation and flushing.
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Functions
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GST_AUDIO_BASE_SRC_CLOCK()
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#define GST_AUDIO_BASE_SRC_CLOCK(obj)    (GST_AUDIO_BASE_SRC (obj)->clock)
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Get the GstClock of obj
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.
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Parameters
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obj
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a GstAudioBaseSrc
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GST_AUDIO_BASE_SRC_PAD()
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#define GST_AUDIO_BASE_SRC_PAD(obj)      (GST_BASE_SRC (obj)->srcpad)
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Get the source GstPad of obj
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.
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Parameters
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obj
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a GstAudioBaseSrc
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gst_audio_base_src_create_ringbuffer ()
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GstAudioRingBuffer *
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gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src);
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Create and return the GstAudioRingBuffer for src
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. This function will call
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the ::create_ringbuffer vmethod and will set src
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 as the parent of the
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returned buffer (see gst_object_set_parent()).
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Parameters
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src
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a GstAudioBaseSrc.
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Returns
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The new ringbuffer of src
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.
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[transfer none]
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gst_audio_base_src_set_provide_clock ()
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void
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gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src,
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                                      gboolean provide);
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Controls whether src
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 will provide a clock or not. If provide
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 is TRUE,
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gst_element_provide_clock() will return a clock that reflects the datarate
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of src
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. If provide
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 is FALSE, gst_element_provide_clock() will return NULL.
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Parameters
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src
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a GstAudioBaseSrc
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provide
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new state
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gst_audio_base_src_get_provide_clock ()
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gboolean
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gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src);
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Queries whether src
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 will provide a clock or not. See also
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gst_audio_base_src_set_provide_clock.
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Parameters
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src
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a GstAudioBaseSrc
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Returns
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 TRUE if src
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will provide a clock.
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gst_audio_base_src_get_slave_method ()
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GstAudioBaseSrcSlaveMethod
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gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src);
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Get the current slave method used by src
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.
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Parameters
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src
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a GstAudioBaseSrc
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Returns
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 The current slave method used by src
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.
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gst_audio_base_src_set_slave_method ()
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void
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gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
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                                     GstAudioBaseSrcSlaveMethod method);
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Controls how clock slaving will be performed in src
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.
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Parameters
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src
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a GstAudioBaseSrc
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method
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the new slave method
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Types and Values
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struct GstAudioBaseSrc
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struct GstAudioBaseSrc;
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Opaque GstAudioBaseSrc.
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struct GstAudioBaseSrcClass
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struct GstAudioBaseSrcClass {
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  GstPushSrcClass      parent_class;
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  /* subclass ringbuffer allocation */
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  GstAudioRingBuffer* (*create_ringbuffer)  (GstAudioBaseSrc *src);
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};
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GstAudioBaseSrc class. Override the vmethod to implement
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functionality.
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Members
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create_ringbuffer ()
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create and return a GstAudioRingBuffer to read from.
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enum GstAudioBaseSrcSlaveMethod
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Different possible clock slaving algorithms when the internal audio clock was
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not selected as the pipeline clock.
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Members
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GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE
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Resample to match the master clock.
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GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP
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Retimestamp output buffers with master
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clock time.
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GST_AUDIO_BASE_SRC_SLAVE_SKEW
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Adjust capture pointer when master clock
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drifts too much.
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GST_AUDIO_BASE_SRC_SLAVE_NONE
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No adjustment is done.
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Property Details
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The “actual-buffer-time” property
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  “actual-buffer-time”       gint64
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Actual configured size of audio buffer in microseconds.
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Flags: Read
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Allowed values: >= -1
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Default value: -1
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The “actual-latency-time” property
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  “actual-latency-time”      gint64
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Actual configured audio latency in microseconds.
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Flags: Read
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Allowed values: >= -1
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Default value: -1
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The “buffer-time” property
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  “buffer-time”              gint64
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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. This value might be ignored by the element if necessary; see "actual-buffer-time".
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Flags: Read / Write
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Allowed values: >= 1
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Default value: 200000
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The “latency-time” property
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  “latency-time”             gint64
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The minimum amount of data to read in each iteration in microseconds. This is the minimum latency that the source reports. This value might be ignored by the element if necessary; see "actual-latency-time".
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Flags: Read / Write
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Allowed values: >= 1
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Default value: 10000
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The “provide-clock” property
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  “provide-clock”            gboolean
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Provide a clock to be used as the global pipeline clock.
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Flags: Read / Write
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Default value: TRUE
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The “slave-method” property
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  “slave-method”             GstAudioBaseSrcSlaveMethod
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Algorithm used to match the rate of the masterclock.
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Flags: Read / Write
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Default value: GST_AUDIO_BASE_SRC_SLAVE_SKEW
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See Also
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GstAudioSrc, GstAudioRingBuffer.
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