gstaudiobasesink: GStreamer Base Plugins 1.0 Library Reference Manual

## gstaudiobasesink gstaudiobasesink — Base class for audio sinks |

## Functions

void | (*GstAudioBaseSinkCustomSlavingCallback) ()
#define | GST_AUDIO_BASE_SINK_CLOCK()
#define | GST_AUDIO_BASE_SINK_PAD()
GstAudioRingBuffer * | gst_audio_base_sink_create_ringbuffer ()
void | gst_audio_base_sink_set_provide_clock ()
gboolean | gst_audio_base_sink_get_provide_clock ()
void | gst_audio_base_sink_set_slave_method ()
GstAudioBaseSinkSlaveMethod | gst_audio_base_sink_get_slave_method ()
void | gst_audio_base_sink_set_drift_tolerance ()
gint64 | gst_audio_base_sink_get_drift_tolerance ()
void | gst_audio_base_sink_set_alignment_threshold ()
GstClockTime | gst_audio_base_sink_get_alignment_threshold ()
void | gst_audio_base_sink_set_discont_wait ()
GstClockTime | gst_audio_base_sink_get_discont_wait ()
void | gst_audio_base_sink_set_custom_slaving_callback ()
void | gst_audio_base_sink_report_device_failure ()

## Properties

guint64 | alignment-threshold | Read / Write
gint64 | buffer-time | Read / Write
gboolean | can-activate-pull | Read / Write
guint64 | discont-wait | Read / Write
gint64 | drift-tolerance | Read / Write
gint64 | latency-time | Read / Write
gboolean | provide-clock | Read / Write
GstAudioBaseSinkSlaveMethod | slave-method | Read / Write

## Types and Values

struct | GstAudioBaseSink
struct | GstAudioBaseSinkClass
enum | GstAudioBaseSinkSlaveMethod
enum | GstAudioBaseSinkDiscontReason

## 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

`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.