GstAudioEncoder

GstAudio.AudioEncoder

GObject.Object
    ╰──GObject.InitiallyUnowned
        ╰──Gst.Object
            ╰──Gst.Element
                ╰──GstAudio.AudioEncoder

GstAudio.AudioEncoder.prototype.allocate_output_buffer

function GstAudio.AudioEncoder.prototype.allocate_output_buffer(size: Number): {
    // javascript wrapper for 'gst_audio_encoder_allocate_output_buffer'
}

Helper function that allocates a buffer to hold an encoded audio frame for enc's current output format.

GstAudio.AudioEncoder.prototype.finish_frame

function GstAudio.AudioEncoder.prototype.finish_frame(buffer: Gst.Buffer, samples: Number): {
    // javascript wrapper for 'gst_audio_encoder_finish_frame'
}

Collects encoded data and pushes encoded data downstream. Source pad caps must be set when this is called.

If samples < 0, then best estimate is all samples provided to encoder (subclass) so far. buf may be NULL, in which case next number of samples are considered discarded, e.g. as a result of discontinuous transmission, and a discontinuity is marked.

Note that samples received in vfunc_handle_frame() may be invalidated by a call to this function.

GstAudio.AudioEncoder.prototype.get_allocator

function GstAudio.AudioEncoder.prototype.get_allocator(): {
    // javascript wrapper for 'gst_audio_encoder_get_allocator'
}

Lets GstAudio.AudioEncoder sub-classes to know the memory allocator used by the base class and its params.

Unref the allocator after use it.

GstAudio.AudioEncoder.prototype.get_audio_info

function GstAudio.AudioEncoder.prototype.get_audio_info(): {
    // javascript wrapper for 'gst_audio_encoder_get_audio_info'
}

GstAudio.AudioEncoder.prototype.get_drainable

function GstAudio.AudioEncoder.prototype.get_drainable(): {
    // javascript wrapper for 'gst_audio_encoder_get_drainable'
}

Queries encoder drain handling.

GstAudio.AudioEncoder.prototype.get_frame_max

function GstAudio.AudioEncoder.prototype.get_frame_max(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_max'
}

GstAudio.AudioEncoder.prototype.get_frame_samples_max

function GstAudio.AudioEncoder.prototype.get_frame_samples_max(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_samples_max'
}

GstAudio.AudioEncoder.prototype.get_frame_samples_min

function GstAudio.AudioEncoder.prototype.get_frame_samples_min(): {
    // javascript wrapper for 'gst_audio_encoder_get_frame_samples_min'
}

GstAudio.AudioEncoder.prototype.get_hard_min

function GstAudio.AudioEncoder.prototype.get_hard_min(): {
    // javascript wrapper for 'gst_audio_encoder_get_hard_min'
}

Queries encoder hard minimum handling.

GstAudio.AudioEncoder.prototype.get_hard_resync

function GstAudio.AudioEncoder.prototype.get_hard_resync(): {
    // javascript wrapper for 'gst_audio_encoder_get_hard_resync'
}

GstAudio.AudioEncoder.prototype.get_latency

function GstAudio.AudioEncoder.prototype.get_latency(): {
    // javascript wrapper for 'gst_audio_encoder_get_latency'
}

Sets the variables pointed to by min and max to the currently configured latency.

GstAudio.AudioEncoder.prototype.get_lookahead

function GstAudio.AudioEncoder.prototype.get_lookahead(): {
    // javascript wrapper for 'gst_audio_encoder_get_lookahead'
}

GstAudio.AudioEncoder.prototype.get_mark_granule

function GstAudio.AudioEncoder.prototype.get_mark_granule(): {
    // javascript wrapper for 'gst_audio_encoder_get_mark_granule'
}

Queries if the encoder will handle granule marking.

GstAudio.AudioEncoder.prototype.get_perfect_timestamp

function GstAudio.AudioEncoder.prototype.get_perfect_timestamp(): {
    // javascript wrapper for 'gst_audio_encoder_get_perfect_timestamp'
}

Queries encoder perfect timestamp behaviour.

GstAudio.AudioEncoder.prototype.get_tolerance

function GstAudio.AudioEncoder.prototype.get_tolerance(): {
    // javascript wrapper for 'gst_audio_encoder_get_tolerance'
}

Queries current audio jitter tolerance threshold.

GstAudio.AudioEncoder.prototype.merge_tags

function GstAudio.AudioEncoder.prototype.merge_tags(tags: Gst.TagList, mode: Gst.TagMergeMode): {
    // javascript wrapper for 'gst_audio_encoder_merge_tags'
}

Sets the audio encoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with GstAudio.AudioEncoder.prototype.merge_tags.

Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.

MT safe.

GstAudio.AudioEncoder.prototype.negotiate

function GstAudio.AudioEncoder.prototype.negotiate(): {
    // javascript wrapper for 'gst_audio_encoder_negotiate'
}

Negotiate with downstream elements to currently configured Gst.Caps. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.

GstAudio.AudioEncoder.prototype.proxy_getcaps

function GstAudio.AudioEncoder.prototype.proxy_getcaps(caps: Gst.Caps, filter: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_proxy_getcaps'
}

Returns caps that express caps (or sink template caps if caps == NULL) restricted to channel/rate combinations supported by downstream elements (e.g. muxers).

GstAudio.AudioEncoder.prototype.set_allocation_caps

function GstAudio.AudioEncoder.prototype.set_allocation_caps(allocation_caps: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_set_allocation_caps'
}

Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling GstAudio.AudioEncoder.prototype.negotiate. Setting to null the allocation query will use the caps from the pad.

Since : 1.10

GstAudio.AudioEncoder.prototype.set_drainable

function GstAudio.AudioEncoder.prototype.set_drainable(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_drainable'
}

Configures encoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover encoded data. Otherwise, it is not considered so capable and will only ever be passed real data.

MT safe.

GstAudio.AudioEncoder.prototype.set_frame_max

function GstAudio.AudioEncoder.prototype.set_frame_max(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_max'
}

Sets max number of frames accepted at once (assumed minimally 1). Requires frame_samples_min and frame_samples_max to be the equal.

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

GstAudio.AudioEncoder.prototype.set_frame_samples_max

function GstAudio.AudioEncoder.prototype.set_frame_samples_max(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_samples_max'
}

Sets number of samples (per channel) subclass needs to be handed, at most or will be handed all available if 0.

If an exact number of samples is required, GstAudio.AudioEncoder.prototype.set_frame_samples_min must be called with the same number.

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

GstAudio.AudioEncoder.prototype.set_frame_samples_min

function GstAudio.AudioEncoder.prototype.set_frame_samples_min(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_frame_samples_min'
}

Sets number of samples (per channel) subclass needs to be handed, at least or will be handed all available if 0.

If an exact number of samples is required, GstAudio.AudioEncoder.prototype.set_frame_samples_max must be called with the same number.

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

GstAudio.AudioEncoder.prototype.set_hard_min

function GstAudio.AudioEncoder.prototype.set_hard_min(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_hard_min'
}

Configures encoder hard minimum handling. If enabled, subclass will never be handed less samples than it configured, which otherwise might occur near end-of-data handling. Instead, the leftover samples will simply be discarded.

MT safe.

GstAudio.AudioEncoder.prototype.set_hard_resync

function GstAudio.AudioEncoder.prototype.set_hard_resync(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_hard_resync'
}

GstAudio.AudioEncoder.prototype.set_headers

function GstAudio.AudioEncoder.prototype.set_headers(headers: [ Gst.Buffer ]): {
    // javascript wrapper for 'gst_audio_encoder_set_headers'
}

Set the codec headers to be sent downstream whenever requested.

GstAudio.AudioEncoder.prototype.set_latency

function GstAudio.AudioEncoder.prototype.set_latency(min: Number, max: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_latency'
}

Sets encoder latency. If the provided values changed from previously provided ones, this will also post a LATENCY message on the bus so the pipeline can reconfigure its global latency.

GstAudio.AudioEncoder.prototype.set_lookahead

function GstAudio.AudioEncoder.prototype.set_lookahead(num: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_lookahead'
}

Sets encoder lookahead (in units of input rate samples)

Note: This value will be reset to 0 every time before vfunc_set_format() is called.

GstAudio.AudioEncoder.prototype.set_mark_granule

function GstAudio.AudioEncoder.prototype.set_mark_granule(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_mark_granule'
}

Enable or disable encoder granule handling.

MT safe.

GstAudio.AudioEncoder.prototype.set_output_format

function GstAudio.AudioEncoder.prototype.set_output_format(caps: Gst.Caps): {
    // javascript wrapper for 'gst_audio_encoder_set_output_format'
}

Configure output caps on the srcpad of enc.

GstAudio.AudioEncoder.prototype.set_perfect_timestamp

function GstAudio.AudioEncoder.prototype.set_perfect_timestamp(enabled: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_perfect_timestamp'
}

Enable or disable encoder perfect output timestamp preference.

MT safe.

GstAudio.AudioEncoder.prototype.set_tolerance

function GstAudio.AudioEncoder.prototype.set_tolerance(tolerance: Number): {
    // javascript wrapper for 'gst_audio_encoder_set_tolerance'
}

Configures encoder audio jitter tolerance threshold.

MT safe.

hard-resync

“hard-resync” Number

Flags : Read / Write

mark-granule

“mark-granule” Number

Flags : Read

perfect-timestamp

“perfect-timestamp” Number

Flags : Read / Write

tolerance

“tolerance” Number

Flags : Read / Write

vfunc_close

function vfunc_close(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'close' virtual method
}

Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources.

vfunc_decide_allocation

function vfunc_decide_allocation(enc: GstAudio.AudioEncoder, query: Gst.Query): {
    // javascript implementation of the 'decide_allocation' virtual method
}

Optional. Setup the allocation parameters for allocating output buffers. The passed in query contains the result of the downstream allocation query. Subclasses should chain up to the parent implementation to invoke the default handler.

vfunc_flush

function vfunc_flush(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'flush' virtual method
}

Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned encoded data.

vfunc_getcaps

function vfunc_getcaps(enc: GstAudio.AudioEncoder, filter: Gst.Caps): {
    // javascript implementation of the 'getcaps' virtual method
}

Optional. Allows for a custom sink getcaps implementation (e.g. for multichannel input specification). If not implemented, default returns gst_audio_encoder_proxy_getcaps applied to sink template caps.

vfunc_handle_frame

function vfunc_handle_frame(enc: GstAudio.AudioEncoder, buffer: Gst.Buffer): {
    // javascript implementation of the 'handle_frame' virtual method
}

Provides input samples (or NULL to clear any remaining data) according to directions as configured by the subclass using the API. Input data ref management is performed by base class, subclass should not care or intervene, and input data is only valid until next call to base class, most notably a call to GstAudio.AudioEncoder.prototype.finish_frame.

vfunc_negotiate

function vfunc_negotiate(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'negotiate' virtual method
}

Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler.

vfunc_open

function vfunc_open(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'open' virtual method
}

Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources.

vfunc_pre_push

function vfunc_pre_push(enc: GstAudio.AudioEncoder, buffer: Gst.Buffer): {
    // javascript implementation of the 'pre_push' virtual method
}

Optional. Called just prior to pushing (encoded data) buffer downstream. Subclass has full discretionary access to buffer, and a not OK flow return will abort downstream pushing.

vfunc_propose_allocation

function vfunc_propose_allocation(enc: GstAudio.AudioEncoder, query: Gst.Query): {
    // javascript implementation of the 'propose_allocation' virtual method
}

Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler.

vfunc_set_format

function vfunc_set_format(enc: GstAudio.AudioEncoder, info: GstAudio.AudioInfo): {
    // javascript implementation of the 'set_format' virtual method
}

Notifies subclass of incoming data format. GstAudioInfo contains the format according to provided caps.

vfunc_sink_event

function vfunc_sink_event(enc: GstAudio.AudioEncoder, event: Gst.Event): {
    // javascript implementation of the 'sink_event' virtual method
}

Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler.

vfunc_sink_query

function vfunc_sink_query(encoder: GstAudio.AudioEncoder, query: Gst.Query): {
    // javascript implementation of the 'sink_query' virtual method
}

Optional. Query handler on the sink pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6

vfunc_src_event

function vfunc_src_event(enc: GstAudio.AudioEncoder, event: Gst.Event): {
    // javascript implementation of the 'src_event' virtual method
}

Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler.

vfunc_src_query

function vfunc_src_query(encoder: GstAudio.AudioEncoder, query: Gst.Query): {
    // javascript implementation of the 'src_query' virtual method
}

Optional. Query handler on the source pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since: 1.6

vfunc_start

function vfunc_start(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'start' virtual method
}

Optional. Called when the element starts processing. Allows opening external resources.

vfunc_stop

function vfunc_stop(enc: GstAudio.AudioEncoder): {
    // javascript implementation of the 'stop' virtual method
}

Optional. Called when the element stops processing. Allows closing external resources.

vfunc_transform_meta

function vfunc_transform_meta(enc: GstAudio.AudioEncoder, outbuf: Gst.Buffer, meta: Gst.Meta, inbuf: Gst.Buffer): {
    // javascript implementation of the 'transform_meta' virtual method
}

Optional. Transform the metadata on the input buffer to the output buffer. By default this method copies all meta without tags and meta with only the "audio" tag. subclasses can implement this method and return true if the metadata is to be copied. Since: 1.6

GstAudio.AUDIO_ENCODER_SINK_NAME

the name of the templates for the sink pad

GstAudio.AUDIO_ENCODER_SRC_NAME

the name of the templates for the source pad