GstAudioBaseSink

GstAudio.AudioBaseSink

GObject.Object
    ╰──GObject.InitiallyUnowned
        ╰──Gst.Object
            ╰──Gst.Element
                ╰──GstBase.BaseSink
                    ╰──GstAudio.AudioBaseSink
                        ╰──GstAudio.AudioSink

GstAudio.AudioBaseSink.prototype.create_ringbuffer

function GstAudio.AudioBaseSink.prototype.create_ringbuffer(): {
    // javascript wrapper for 'gst_audio_base_sink_create_ringbuffer'
}

Create and return the GstAudio.AudioRingBuffer for sink. This function will call the ::create_ringbuffer vmethod and will set sink as the parent of the returned buffer (see Gst.Object.prototype.set_parent).

GstAudio.AudioBaseSink.prototype.get_alignment_threshold

function GstAudio.AudioBaseSink.prototype.get_alignment_threshold(): {
    // javascript wrapper for 'gst_audio_base_sink_get_alignment_threshold'
}

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

GstAudio.AudioBaseSink.prototype.get_discont_wait

function GstAudio.AudioBaseSink.prototype.get_discont_wait(): {
    // javascript wrapper for 'gst_audio_base_sink_get_discont_wait'
}

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

GstAudio.AudioBaseSink.prototype.get_drift_tolerance

function GstAudio.AudioBaseSink.prototype.get_drift_tolerance(): {
    // javascript wrapper for 'gst_audio_base_sink_get_drift_tolerance'
}

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

GstAudio.AudioBaseSink.prototype.get_provide_clock

function GstAudio.AudioBaseSink.prototype.get_provide_clock(): {
    // javascript wrapper for 'gst_audio_base_sink_get_provide_clock'
}

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

GstAudio.AudioBaseSink.prototype.get_slave_method

function GstAudio.AudioBaseSink.prototype.get_slave_method(): {
    // javascript wrapper for 'gst_audio_base_sink_get_slave_method'
}

Get the current slave method used by sink.

GstAudio.AudioBaseSink.prototype.report_device_failure

function GstAudio.AudioBaseSink.prototype.report_device_failure(): {
    // javascript wrapper for 'gst_audio_base_sink_report_device_failure'
}

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.

Since : 1.6

GstAudio.AudioBaseSink.prototype.set_alignment_threshold

function GstAudio.AudioBaseSink.prototype.set_alignment_threshold(alignment_threshold: Number): {
    // javascript wrapper for 'gst_audio_base_sink_set_alignment_threshold'
}

Controls the sink's alignment threshold.

GstAudio.AudioBaseSink.prototype.set_custom_slaving_callback

function GstAudio.AudioBaseSink.prototype.set_custom_slaving_callback(callback: GstAudio.AudioBaseSinkCustomSlavingCallback, user_data: Object): {
    // javascript wrapper for 'gst_audio_base_sink_set_custom_slaving_callback'
}

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.

Since : 1.6

GstAudio.AudioBaseSink.prototype.set_discont_wait

function GstAudio.AudioBaseSink.prototype.set_discont_wait(discont_wait: Number): {
    // javascript wrapper for 'gst_audio_base_sink_set_discont_wait'
}

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

GstAudio.AudioBaseSink.prototype.set_drift_tolerance

function GstAudio.AudioBaseSink.prototype.set_drift_tolerance(drift_tolerance: Number): {
    // javascript wrapper for 'gst_audio_base_sink_set_drift_tolerance'
}

Controls the sink's drift tolerance.

GstAudio.AudioBaseSink.prototype.set_provide_clock

function GstAudio.AudioBaseSink.prototype.set_provide_clock(provide: Number): {
    // javascript wrapper for 'gst_audio_base_sink_set_provide_clock'
}

Controls whether sink will provide a clock or not. If provide is true, Gst.Element.prototype.provide_clock will return a clock that reflects the datarate of sink. If provide is false, Gst.Element.prototype.provide_clock will return NULL.

GstAudio.AudioBaseSink.prototype.set_slave_method

function GstAudio.AudioBaseSink.prototype.set_slave_method(method: GstAudio.AudioBaseSinkSlaveMethod): {
    // javascript wrapper for 'gst_audio_base_sink_set_slave_method'
}

Controls how clock slaving will be performed in sink.

alignment-threshold

“alignment-threshold” Number

Flags : Read / Write

buffer-time

“buffer-time” Number

Flags : Read / Write

can-activate-pull

“can-activate-pull” Number

Flags : Read / Write

discont-wait

“discont-wait” Number

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

Flags : Read / Write

drift-tolerance

“drift-tolerance” Number

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

Flags : Read / Write

latency-time

“latency-time” Number

Flags : Read / Write

provide-clock

“provide-clock” Number

Flags : Read / Write

slave-method

“slave-method” GstAudio.AudioBaseSinkSlaveMethod

Flags : Read / Write

vfunc_create_ringbuffer

function vfunc_create_ringbuffer(sink: GstAudio.AudioBaseSink): {
    // javascript implementation of the 'create_ringbuffer' virtual method
}

create and return a GstAudio.AudioRingBuffer to write to.

vfunc_payload

function vfunc_payload(sink: GstAudio.AudioBaseSink, buffer: Gst.Buffer): {
    // javascript implementation of the 'payload' virtual method
}

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.

GstAudio.AudioBaseSinkDiscontReason

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

Since : 1.6

GstAudio.AudioBaseSinkSlaveMethod

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

GstAudio.AudioBaseSinkCustomSlavingCallback

function GstAudio.AudioBaseSinkCustomSlavingCallback(sink: GstAudio.AudioBaseSink, etime: Number, itime: Number, requested_skew: Number, discont_reason: GstAudio.AudioBaseSinkDiscontReason, user_data: Object): {
    // javascript wrapper for 'GstAudioBaseSinkCustomSlavingCallback'
}

This function is set with GstAudio.AudioBaseSink.prototype.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.

Since : 1.6