gstaudiobasesink

gstaudiobasesink — Base class for audio sinks

Functions

Properties

Types and Values

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstBaseSink
                    ╰── GstAudioBaseSink
                        ╰── GstAudioSink

Includes

#include <gst/audio/gstaudiobasesink.h>

Description

This is the base class for audio sinks. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of writing samples to the ringbuffer, synchronisation, clipping and flushing.

Functions

GstAudioBaseSinkCustomSlavingCallback ()

void
(*GstAudioBaseSinkCustomSlavingCallback)
                               (GstAudioBaseSink *sink,
                                GstClockTime etime,
                                GstClockTime itime,
                                GstClockTimeDiff *requested_skew,
                                GstAudioBaseSinkDiscontReason discont_reason,
                                gpointer user_data);

This function is set with gst_audio_base_sink_set_custom_slaving_callback() and is called during playback. It receives the current time of external and internal clocks, which the callback can then use to apply any custom slaving/synchronization schemes.

The external clock is the sink's element clock, the internal one is the internal audio clock. The internal audio clock's calibration is applied to the timestamps before they are passed to the callback. The difference between etime and itime is the skew; how much internal and external clock lie apart from each other. A skew of 0 means both clocks are perfectly in sync. itime > etime means the external clock is going slower, while itime < etime means it is going faster than the internal clock. etime and itime are always valid timestamps, except for when a discontinuity happens.

requested_skew is an output value the callback can write to. It informs the sink of whether or not it should move the playout pointer, and if so, by how much. This pointer is only NULL if a discontinuity occurs; otherwise, it is safe to write to *requested_skew. The default skew is 0.

The sink may experience discontinuities. If one happens, discont is TRUE, itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. This makes it possible to reset custom clock slaving algorithms when a discontinuity happens.

Parameters

sink

a GstAudioBaseSink

 

etime

external clock time

 

itime

internal clock time

 

requested_skew

skew amount requested by the callback

 

discont_reason

reason for discontinuity (if any)

 

user_data

user data

 

Since: 1.6


GST_AUDIO_BASE_SINK_CLOCK()

#define GST_AUDIO_BASE_SINK_CLOCK(obj)   (GST_AUDIO_BASE_SINK (obj)->clock)

Get the GstClock of obj .

Parameters

obj

a GstAudioBaseSink

 

GST_AUDIO_BASE_SINK_PAD()

#define GST_AUDIO_BASE_SINK_PAD(obj)     (GST_BASE_SINK (obj)->sinkpad)

Get the sink GstPad of obj .

Parameters

obj

a GstAudioBaseSink

 

gst_audio_base_sink_create_ringbuffer ()

GstAudioRingBuffer *
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);

Create and return the GstAudioRingBuffer for sink . This function will call the ::create_ringbuffer vmethod and will set sink as the parent of the returned buffer (see gst_object_set_parent()).

Parameters

sink

a GstAudioBaseSink.

 

Returns

The new ringbuffer of sink .

[transfer none]


gst_audio_base_sink_set_provide_clock ()

void
gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink,
                                       gboolean provide);

Controls whether sink will provide a clock or not. If provide is TRUE, gst_element_provide_clock() will return a clock that reflects the datarate of sink . If provide is FALSE, gst_element_provide_clock() will return NULL.

Parameters

sink

a GstAudioBaseSink

 

provide

new state

 

gst_audio_base_sink_get_provide_clock ()

gboolean
gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);

Queries whether sink will provide a clock or not. See also gst_audio_base_sink_set_provide_clock.

Parameters

sink

a GstAudioBaseSink

 

Returns

TRUE if sink will provide a clock.


gst_audio_base_sink_set_slave_method ()

void
gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
                                      GstAudioBaseSinkSlaveMethod method);

Controls how clock slaving will be performed in sink .

Parameters

sink

a GstAudioBaseSink

 

method

the new slave method

 

gst_audio_base_sink_get_slave_method ()

GstAudioBaseSinkSlaveMethod
gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);

Get the current slave method used by sink .

Parameters

sink

a GstAudioBaseSink

 

Returns

The current slave method used by sink .


gst_audio_base_sink_set_drift_tolerance ()

void
gst_audio_base_sink_set_drift_tolerance
                               (GstAudioBaseSink *sink,
                                gint64 drift_tolerance);

Controls the sink's drift tolerance.

Parameters

sink

a GstAudioBaseSink

 

drift_tolerance

the new drift tolerance in microseconds

 

gst_audio_base_sink_get_drift_tolerance ()

gint64
gst_audio_base_sink_get_drift_tolerance
                               (GstAudioBaseSink *sink);

Get the current drift tolerance, in microseconds, used by sink .

Parameters

sink

a GstAudioBaseSink

 

Returns

The current drift tolerance used by sink .


gst_audio_base_sink_set_alignment_threshold ()

void
gst_audio_base_sink_set_alignment_threshold
                               (GstAudioBaseSink *sink,
                                GstClockTime alignment_threshold);

Controls the sink's alignment threshold.

Parameters

sink

a GstAudioBaseSink

 

alignment_threshold

the new alignment threshold in nanoseconds

 

gst_audio_base_sink_get_alignment_threshold ()

GstClockTime
gst_audio_base_sink_get_alignment_threshold
                               (GstAudioBaseSink *sink);

Get the current alignment threshold, in nanoseconds, used by sink .

Parameters

sink

a GstAudioBaseSink

 

Returns

The current alignment threshold used by sink .


gst_audio_base_sink_set_discont_wait ()

void
gst_audio_base_sink_set_discont_wait (GstAudioBaseSink *sink,
                                      GstClockTime discont_wait);

Controls how long the sink will wait before creating a discontinuity.

Parameters

sink

a GstAudioBaseSink

 

discont_wait

the new discont wait in nanoseconds

 

gst_audio_base_sink_get_discont_wait ()

GstClockTime
gst_audio_base_sink_get_discont_wait (GstAudioBaseSink *sink);

Get the current discont wait, in nanoseconds, used by sink .

Parameters

sink

a GstAudioBaseSink

 

Returns

The current discont wait used by sink .


gst_audio_base_sink_set_custom_slaving_callback ()

void
gst_audio_base_sink_set_custom_slaving_callback
                               (GstAudioBaseSink *sink,
                                GstAudioBaseSinkCustomSlavingCallback callback,
                                gpointer user_data,
                                GDestroyNotify notify);

Sets the custom slaving callback. This callback will be invoked if the slave-method property is set to GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink receives and plays samples.

Setting the callback to NULL causes the sink to behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE method were used.

Parameters

sink

a GstAudioBaseSink

 

callback

a GstAudioBaseSinkCustomSlavingCallback

 

user_data

user data passed to the callback

 

notify

called when user_data becomes unused

 

Since: 1.6


gst_audio_base_sink_report_device_failure ()

void
gst_audio_base_sink_report_device_failure
                               (GstAudioBaseSink *sink);

Informs this base class that the audio output device has failed for some reason, causing a discontinuity (for example, because the device recovered from the error, but lost all contents of its ring buffer). This function is typically called by derived classes, and is useful for the custom slave method.

Parameters

sink

a GstAudioBaseSink

 

Since: 1.6

Types and Values

struct GstAudioBaseSink

struct GstAudioBaseSink;

Opaque GstAudioBaseSink.


struct GstAudioBaseSinkClass

struct GstAudioBaseSinkClass {
  GstBaseSinkClass     parent_class;

  /* subclass ringbuffer allocation */
  GstAudioRingBuffer* (*create_ringbuffer)  (GstAudioBaseSink *sink);

  /* subclass payloader */
  GstBuffer*          (*payload)            (GstAudioBaseSink *sink,
                                             GstBuffer        *buffer);
};

GstAudioBaseSink class. Override the vmethod to implement functionality.

Members

GstBaseSinkClass parent_class;

the parent class.

 

create_ringbuffer ()

create and return a GstAudioRingBuffer to write to.

 

payload ()

payload data in a format suitable to write to the sink. If no payloading is required, returns a reffed copy of the original buffer, else returns the payloaded buffer with all other metadata copied.

 

enum GstAudioBaseSinkSlaveMethod

Different possible clock slaving algorithms used when the internal audio clock is not selected as the pipeline master clock.

Members

GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE

Resample to match the master clock

 

GST_AUDIO_BASE_SINK_SLAVE_SKEW

Adjust playout pointer when master clock drifts too much.

 

GST_AUDIO_BASE_SINK_SLAVE_NONE

No adjustment is done.

 

GST_AUDIO_BASE_SINK_SLAVE_CUSTOM

Use custom clock slaving algorithm (Since: 1.6)

 

enum GstAudioBaseSinkDiscontReason

Different possible reasons for discontinuities. This enum is useful for the custom slave method.

Members

GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT

No discontinuity occurred

 

GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS

New caps are set, causing renegotiotion

 

GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH

Samples have been flushed

 

GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY

Sink was synchronized to the estimated latency (occurs during initialization)

 

GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT

Aligning buffers failed because the timestamps are too discontinuous

 

GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE

Audio output device experienced and recovered from an error but introduced latency in the process (see also gst_audio_base_sink_report_device_failure() )

 

Since: 1.6

Property Details

The “alignment-threshold” property

  “alignment-threshold”      guint64

Timestamp alignment threshold in nanoseconds.

Flags: Read / Write

Allowed values: [1,18446744073709551614]

Default value: 40000000


The “buffer-time” property

  “buffer-time”              gint64

Size of audio buffer in microseconds, this is the minimum latency that the sink reports.

Flags: Read / Write

Allowed values: >= 1

Default value: 200000


The “can-activate-pull” property

  “can-activate-pull”        gboolean

Allow pull-based scheduling.

Flags: Read / Write

Default value: FALSE


The “discont-wait” property

  “discont-wait”             guint64

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

Flags: Read / Write

Allowed values: <= 18446744073709551614

Default value: 1000000000


The “drift-tolerance” property

  “drift-tolerance”          gint64

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

Flags: Read / Write

Allowed values: >= 1

Default value: 40000


The “latency-time” property

  “latency-time”             gint64

The minimum amount of data to write in each iteration in microseconds.

Flags: Read / Write

Allowed values: >= 1

Default value: 10000


The “provide-clock” property

  “provide-clock”            gboolean

Provide a clock to be used as the global pipeline clock.

Flags: Read / Write

Default value: TRUE


The “slave-method” property

  “slave-method”             GstAudioBaseSinkSlaveMethod

Algorithm used to match the rate of the masterclock.

Flags: Read / Write

Default value: GST_AUDIO_BASE_SINK_SLAVE_SKEW

See Also

GstAudioSink, GstAudioRingBuffer.