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dotfiles/config/ags/quickactions/@girs/gstaudio-1.0.d.ts
Janis Hutz b5ad7e3034 [Notes] Some changes
2025-04-25 06:53:17 +02:00

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/// <reference path="./gstbase-1.0.d.ts" />
/// <reference path="./gst-1.0.d.ts" />
/// <reference path="./gobject-2.0.d.ts" />
/// <reference path="./glib-2.0.d.ts" />
/// <reference path="./gmodule-2.0.d.ts" />
/**
* Type Definitions for Gjs (https://gjs.guide/)
*
* These type definitions are automatically generated, do not edit them by hand.
* If you found a bug fix it in `ts-for-gir` or create a bug report on https://github.com/gjsify/ts-for-gir
*
* The based EJS template file is used for the generated .d.ts file of each GIR module like Gtk-4.0, GObject-2.0, ...
*/
declare module 'gi://GstAudio?version=1.0' {
// Module dependencies
import type GstBase from 'gi://GstBase?version=1.0';
import type Gst from 'gi://Gst?version=1.0';
import type GObject from 'gi://GObject?version=2.0';
import type GLib from 'gi://GLib?version=2.0';
import type GModule from 'gi://GModule?version=2.0';
export namespace GstAudio {
/**
* GstAudio-1.0
*/
/**
* Different possible reasons for discontinuities. This enum is useful for the custom
* slave method.
*/
/**
* Different possible reasons for discontinuities. This enum is useful for the custom
* slave method.
*/
export namespace AudioBaseSinkDiscontReason {
export const $gtype: GObject.GType<AudioBaseSinkDiscontReason>;
}
enum AudioBaseSinkDiscontReason {
/**
* No discontinuity occurred
*/
NO_DISCONT,
/**
* New caps are set, causing renegotiotion
*/
NEW_CAPS,
/**
* Samples have been flushed
*/
FLUSH,
/**
* Sink was synchronized to the estimated latency (occurs during initialization)
*/
SYNC_LATENCY,
/**
* Aligning buffers failed because the timestamps are too discontinuous
*/
ALIGNMENT,
/**
* Audio output device experienced and recovered from an error but introduced latency in the process (see also gst_audio_base_sink_report_device_failure())
*/
DEVICE_FAILURE,
}
/**
* Different possible clock slaving algorithms used when the internal audio
* clock is not selected as the pipeline master clock.
*/
/**
* Different possible clock slaving algorithms used when the internal audio
* clock is not selected as the pipeline master clock.
*/
export namespace AudioBaseSinkSlaveMethod {
export const $gtype: GObject.GType<AudioBaseSinkSlaveMethod>;
}
enum AudioBaseSinkSlaveMethod {
/**
* Resample to match the master clock
*/
RESAMPLE,
/**
* Adjust playout pointer when master clock
* drifts too much.
*/
SKEW,
/**
* No adjustment is done.
*/
NONE,
/**
* Use custom clock slaving algorithm (Since: 1.6)
*/
CUSTOM,
}
/**
* Different possible clock slaving algorithms when the internal audio clock was
* not selected as the pipeline clock.
*/
/**
* Different possible clock slaving algorithms when the internal audio clock was
* not selected as the pipeline clock.
*/
export namespace AudioBaseSrcSlaveMethod {
export const $gtype: GObject.GType<AudioBaseSrcSlaveMethod>;
}
enum AudioBaseSrcSlaveMethod {
/**
* Resample to match the master clock.
*/
RESAMPLE,
/**
* Retimestamp output buffers with master
* clock time.
*/
RE_TIMESTAMP,
/**
* Adjust capture pointer when master clock
* drifts too much.
*/
SKEW,
/**
* No adjustment is done.
*/
NONE,
}
/**
* Mode in which the CD audio source operates. Influences timestamping,
* EOS handling and seeking.
*/
/**
* Mode in which the CD audio source operates. Influences timestamping,
* EOS handling and seeking.
*/
export namespace AudioCdSrcMode {
export const $gtype: GObject.GType<AudioCdSrcMode>;
}
enum AudioCdSrcMode {
/**
* each single track is a stream
*/
NORMAL,
/**
* the entire disc is a single stream
*/
CONTINUOUS,
}
/**
* Audio channel positions.
*
* These are the channels defined in SMPTE 2036-2-2008
* Table 1 for 22.2 audio systems with the Surround and Wide channels from
* DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the
* actual channel layout is expressed with a channel count and a channel mask,
* which describes the existing channels. The positions in the bit mask correspond
* to the enum values.
* For negotiation it is allowed to have more bits set in the channel mask than
* the number of channels to specify the allowed channel positions but this is
* not allowed in negotiated caps. It is not allowed in any situation other
* than the one mentioned below to have less bits set in the channel mask than
* the number of channels.
*
* `GST_AUDIO_CHANNEL_POSITION_MONO` can only be used with a single mono channel that
* has no direction information and would be mixed into all directional channels.
* This is expressed in caps by having a single channel and no channel mask.
*
* `GST_AUDIO_CHANNEL_POSITION_NONE` can only be used if all channels have this position.
* This is expressed in caps by having a channel mask with no bits set.
*
* As another special case it is allowed to have two channels without a channel mask.
* This implicitly means that this is a stereo stream with a front left and front right
* channel.
*/
/**
* Audio channel positions.
*
* These are the channels defined in SMPTE 2036-2-2008
* Table 1 for 22.2 audio systems with the Surround and Wide channels from
* DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the
* actual channel layout is expressed with a channel count and a channel mask,
* which describes the existing channels. The positions in the bit mask correspond
* to the enum values.
* For negotiation it is allowed to have more bits set in the channel mask than
* the number of channels to specify the allowed channel positions but this is
* not allowed in negotiated caps. It is not allowed in any situation other
* than the one mentioned below to have less bits set in the channel mask than
* the number of channels.
*
* `GST_AUDIO_CHANNEL_POSITION_MONO` can only be used with a single mono channel that
* has no direction information and would be mixed into all directional channels.
* This is expressed in caps by having a single channel and no channel mask.
*
* `GST_AUDIO_CHANNEL_POSITION_NONE` can only be used if all channels have this position.
* This is expressed in caps by having a channel mask with no bits set.
*
* As another special case it is allowed to have two channels without a channel mask.
* This implicitly means that this is a stereo stream with a front left and front right
* channel.
*/
export namespace AudioChannelPosition {
export const $gtype: GObject.GType<AudioChannelPosition>;
}
enum AudioChannelPosition {
/**
* used for position-less channels, e.g.
* from a sound card that records 1024 channels; mutually exclusive with
* any other channel position
*/
NONE,
/**
* Mono without direction;
* can only be used with 1 channel
*/
MONO,
/**
* invalid position
*/
INVALID,
/**
* Front left
*/
FRONT_LEFT,
/**
* Front right
*/
FRONT_RIGHT,
/**
* Front center
*/
FRONT_CENTER,
/**
* Low-frequency effects 1 (subwoofer)
*/
LFE1,
/**
* Rear left
*/
REAR_LEFT,
/**
* Rear right
*/
REAR_RIGHT,
/**
* Front left of center
*/
FRONT_LEFT_OF_CENTER,
/**
* Front right of center
*/
FRONT_RIGHT_OF_CENTER,
/**
* Rear center
*/
REAR_CENTER,
/**
* Low-frequency effects 2 (subwoofer)
*/
LFE2,
/**
* Side left
*/
SIDE_LEFT,
/**
* Side right
*/
SIDE_RIGHT,
/**
* Top front left
*/
TOP_FRONT_LEFT,
/**
* Top front right
*/
TOP_FRONT_RIGHT,
/**
* Top front center
*/
TOP_FRONT_CENTER,
/**
* Top center
*/
TOP_CENTER,
/**
* Top rear left
*/
TOP_REAR_LEFT,
/**
* Top rear right
*/
TOP_REAR_RIGHT,
/**
* Top side right
*/
TOP_SIDE_LEFT,
/**
* Top rear right
*/
TOP_SIDE_RIGHT,
/**
* Top rear center
*/
TOP_REAR_CENTER,
/**
* Bottom front center
*/
BOTTOM_FRONT_CENTER,
/**
* Bottom front left
*/
BOTTOM_FRONT_LEFT,
/**
* Bottom front right
*/
BOTTOM_FRONT_RIGHT,
/**
* Wide left (between front left and side left)
*/
WIDE_LEFT,
/**
* Wide right (between front right and side right)
*/
WIDE_RIGHT,
/**
* Surround left (between rear left and side left)
*/
SURROUND_LEFT,
/**
* Surround right (between rear right and side right)
*/
SURROUND_RIGHT,
/**
* Top surround left (between rear left and side left).
*/
TOP_SURROUND_LEFT,
/**
* Top surround right (between rear right and side right).
*/
TOP_SURROUND_RIGHT,
}
/**
* Set of available dithering methods.
*/
/**
* Set of available dithering methods.
*/
export namespace AudioDitherMethod {
export const $gtype: GObject.GType<AudioDitherMethod>;
}
enum AudioDitherMethod {
/**
* No dithering
*/
NONE,
/**
* Rectangular dithering
*/
RPDF,
/**
* Triangular dithering (default)
*/
TPDF,
/**
* High frequency triangular dithering
*/
TPDF_HF,
}
/**
* Enum value describing the most common audio formats.
*/
/**
* Enum value describing the most common audio formats.
*/
export namespace AudioFormat {
export const $gtype: GObject.GType<AudioFormat>;
}
enum AudioFormat {
/**
* unknown or unset audio format
*/
UNKNOWN,
/**
* encoded audio format
*/
ENCODED,
/**
* 8 bits in 8 bits, signed
*/
S8,
/**
* 8 bits in 8 bits, unsigned
*/
U8,
/**
* 16 bits in 16 bits, signed, little endian
*/
S16LE,
/**
* 16 bits in 16 bits, signed, big endian
*/
S16BE,
/**
* 16 bits in 16 bits, unsigned, little endian
*/
U16LE,
/**
* 16 bits in 16 bits, unsigned, big endian
*/
U16BE,
/**
* 24 bits in 32 bits, signed, little endian
*/
S24_32LE,
/**
* 24 bits in 32 bits, signed, big endian
*/
S24_32BE,
/**
* 24 bits in 32 bits, unsigned, little endian
*/
U24_32LE,
/**
* 24 bits in 32 bits, unsigned, big endian
*/
U24_32BE,
/**
* 32 bits in 32 bits, signed, little endian
*/
S32LE,
/**
* 32 bits in 32 bits, signed, big endian
*/
S32BE,
/**
* 32 bits in 32 bits, unsigned, little endian
*/
U32LE,
/**
* 32 bits in 32 bits, unsigned, big endian
*/
U32BE,
/**
* 24 bits in 24 bits, signed, little endian
*/
S24LE,
/**
* 24 bits in 24 bits, signed, big endian
*/
S24BE,
/**
* 24 bits in 24 bits, unsigned, little endian
*/
U24LE,
/**
* 24 bits in 24 bits, unsigned, big endian
*/
U24BE,
/**
* 20 bits in 24 bits, signed, little endian
*/
S20LE,
/**
* 20 bits in 24 bits, signed, big endian
*/
S20BE,
/**
* 20 bits in 24 bits, unsigned, little endian
*/
U20LE,
/**
* 20 bits in 24 bits, unsigned, big endian
*/
U20BE,
/**
* 18 bits in 24 bits, signed, little endian
*/
S18LE,
/**
* 18 bits in 24 bits, signed, big endian
*/
S18BE,
/**
* 18 bits in 24 bits, unsigned, little endian
*/
U18LE,
/**
* 18 bits in 24 bits, unsigned, big endian
*/
U18BE,
/**
* 32-bit floating point samples, little endian
*/
F32LE,
/**
* 32-bit floating point samples, big endian
*/
F32BE,
/**
* 64-bit floating point samples, little endian
*/
F64LE,
/**
* 64-bit floating point samples, big endian
*/
F64BE,
/**
* 16 bits in 16 bits, signed, native endianness
*/
S16,
/**
* 16 bits in 16 bits, unsigned, native endianness
*/
U16,
/**
* 24 bits in 32 bits, signed, native endianness
*/
S24_32,
/**
* 24 bits in 32 bits, unsigned, native endianness
*/
U24_32,
/**
* 32 bits in 32 bits, signed, native endianness
*/
S32,
/**
* 32 bits in 32 bits, unsigned, native endianness
*/
U32,
/**
* 24 bits in 24 bits, signed, native endianness
*/
S24,
/**
* 24 bits in 24 bits, unsigned, native endianness
*/
U24,
/**
* 20 bits in 24 bits, signed, native endianness
*/
S20,
/**
* 20 bits in 24 bits, unsigned, native endianness
*/
U20,
/**
* 18 bits in 24 bits, signed, native endianness
*/
S18,
/**
* 18 bits in 24 bits, unsigned, native endianness
*/
U18,
/**
* 32-bit floating point samples, native endianness
*/
F32,
/**
* 64-bit floating point samples, native endianness
*/
F64,
}
/**
* Layout of the audio samples for the different channels.
*/
/**
* Layout of the audio samples for the different channels.
*/
export namespace AudioLayout {
export const $gtype: GObject.GType<AudioLayout>;
}
enum AudioLayout {
/**
* interleaved audio
*/
INTERLEAVED,
/**
* non-interleaved audio
*/
NON_INTERLEAVED,
}
/**
* Set of available noise shaping methods
*/
/**
* Set of available noise shaping methods
*/
export namespace AudioNoiseShapingMethod {
export const $gtype: GObject.GType<AudioNoiseShapingMethod>;
}
enum AudioNoiseShapingMethod {
/**
* No noise shaping (default)
*/
NONE,
/**
* Error feedback
*/
ERROR_FEEDBACK,
/**
* Simple 2-pole noise shaping
*/
SIMPLE,
/**
* Medium 5-pole noise shaping
*/
MEDIUM,
/**
* High 8-pole noise shaping
*/
HIGH,
}
/**
* The different filter interpolation methods.
*/
/**
* The different filter interpolation methods.
*/
export namespace AudioResamplerFilterInterpolation {
export const $gtype: GObject.GType<AudioResamplerFilterInterpolation>;
}
enum AudioResamplerFilterInterpolation {
/**
* no interpolation
*/
NONE,
/**
* linear interpolation of the
* filter coefficients.
*/
LINEAR,
/**
* cubic interpolation of the
* filter coefficients.
*/
CUBIC,
}
/**
* Select for the filter tables should be set up.
*/
/**
* Select for the filter tables should be set up.
*/
export namespace AudioResamplerFilterMode {
export const $gtype: GObject.GType<AudioResamplerFilterMode>;
}
enum AudioResamplerFilterMode {
/**
* Use interpolated filter tables. This
* uses less memory but more CPU and is slightly less accurate but it allows for more
* efficient variable rate resampling with gst_audio_resampler_update().
*/
INTERPOLATED,
/**
* Use full filter table. This uses more memory
* but less CPU.
*/
FULL,
/**
* Automatically choose between interpolated
* and full filter tables.
*/
AUTO,
}
/**
* Different subsampling and upsampling methods
*/
/**
* Different subsampling and upsampling methods
*/
export namespace AudioResamplerMethod {
export const $gtype: GObject.GType<AudioResamplerMethod>;
}
enum AudioResamplerMethod {
/**
* Duplicates the samples when
* upsampling and drops when downsampling
*/
NEAREST,
/**
* Uses linear interpolation to reconstruct
* missing samples and averaging to downsample
*/
LINEAR,
/**
* Uses cubic interpolation
*/
CUBIC,
/**
* Uses Blackman-Nuttall windowed sinc interpolation
*/
BLACKMAN_NUTTALL,
/**
* Uses Kaiser windowed sinc interpolation
*/
KAISER,
}
/**
* The format of the samples in the ringbuffer.
*/
/**
* The format of the samples in the ringbuffer.
*/
export namespace AudioRingBufferFormatType {
export const $gtype: GObject.GType<AudioRingBufferFormatType>;
}
enum AudioRingBufferFormatType {
/**
* samples in linear or float
*/
RAW,
/**
* samples in mulaw
*/
MU_LAW,
/**
* samples in alaw
*/
A_LAW,
/**
* samples in ima adpcm
*/
IMA_ADPCM,
/**
* samples in mpeg audio (but not AAC) format
*/
MPEG,
/**
* samples in gsm format
*/
GSM,
/**
* samples in IEC958 frames (e.g. AC3)
*/
IEC958,
/**
* samples in AC3 format
*/
AC3,
/**
* samples in EAC3 format
*/
EAC3,
/**
* samples in DTS format
*/
DTS,
/**
* samples in MPEG-2 AAC ADTS format
*/
MPEG2_AAC,
/**
* samples in MPEG-4 AAC ADTS format
*/
MPEG4_AAC,
/**
* samples in MPEG-2 AAC raw format (Since: 1.12)
*/
MPEG2_AAC_RAW,
/**
* samples in MPEG-4 AAC raw format (Since: 1.12)
*/
MPEG4_AAC_RAW,
/**
* samples in FLAC format (Since: 1.12)
*/
FLAC,
/**
* samples in DSD format (Since: 1.24)
*/
DSD,
}
/**
* The state of the ringbuffer.
*/
/**
* The state of the ringbuffer.
*/
export namespace AudioRingBufferState {
export const $gtype: GObject.GType<AudioRingBufferState>;
}
enum AudioRingBufferState {
/**
* The ringbuffer is stopped
*/
STOPPED,
/**
* The ringbuffer is paused
*/
PAUSED,
/**
* The ringbuffer is started
*/
STARTED,
/**
* The ringbuffer has encountered an
* error after it has been started, e.g. because the device was
* disconnected (Since: 1.2)
*/
ERROR,
}
/**
* Enum value describing how DSD bits are grouped.
*/
/**
* Enum value describing how DSD bits are grouped.
*/
export namespace DsdFormat {
export const $gtype: GObject.GType<DsdFormat>;
}
enum DsdFormat {
/**
* unknown / invalid DSD format
*/
DSD_FORMAT_UNKNOWN,
/**
* 8 DSD bits in 1 byte
*/
DSD_FORMAT_U8,
/**
* 16 DSD bits in 2 bytes, little endian order
*/
DSD_FORMAT_U16LE,
/**
* 16 DSD bits in 2 bytes, big endian order
*/
DSD_FORMAT_U16BE,
/**
* 32 DSD bits in 4 bytes, little endian order
*/
DSD_FORMAT_U32LE,
/**
* 32 DSD bits in 4 bytes, big endian order
*/
DSD_FORMAT_U32BE,
/**
* number of valid DSD formats
*/
NUM_DSD_FORMATS,
/**
* 16 DSD bits in 2 bytes, native endianness
*/
DSD_FORMAT_U16,
/**
* 32 DSD bits in 4 bytes, native endianness
*/
DSD_FORMAT_U32,
}
/**
* Different representations of a stream volume. gst_stream_volume_convert_volume()
* allows to convert between the different representations.
*
* Formulas to convert from a linear to a cubic or dB volume are
* cbrt(val) and 20 * log10 (val).
*/
/**
* Different representations of a stream volume. gst_stream_volume_convert_volume()
* allows to convert between the different representations.
*
* Formulas to convert from a linear to a cubic or dB volume are
* cbrt(val) and 20 * log10 (val).
*/
export namespace StreamVolumeFormat {
export const $gtype: GObject.GType<StreamVolumeFormat>;
}
enum StreamVolumeFormat {
/**
* Linear scale factor, 1.0 = 100%
*/
LINEAR,
/**
* Cubic volume scale
*/
CUBIC,
/**
* Logarithmic volume scale (dB, amplitude not power)
*/
DB,
}
/**
* Maximum range of allowed channels, for use in template caps strings.
*/
const AUDIO_CHANNELS_RANGE: string;
/**
* #GstAudioDitherMethod, The dither method to use when
* changing bit depth.
* Default is #GST_AUDIO_DITHER_NONE.
*/
const AUDIO_CONVERTER_OPT_DITHER_METHOD: string;
/**
* Threshold for the output bit depth at/below which to apply dithering.
*
* Default is 20 bit.
*/
const AUDIO_CONVERTER_OPT_DITHER_THRESHOLD: string;
/**
* #GST_TYPE_LIST, The channel mapping matrix.
*
* The matrix coefficients must be between -1 and 1: the number of rows is equal
* to the number of output channels and the number of columns is equal to the
* number of input channels.
*
* ## Example matrix generation code
* To generate the matrix using code:
*
*
* ```
* GValue v = G_VALUE_INIT;
* GValue v2 = G_VALUE_INIT;
* GValue v3 = G_VALUE_INIT;
*
* g_value_init (&v2, GST_TYPE_ARRAY);
* g_value_init (&v3, G_TYPE_DOUBLE);
* g_value_set_double (&v3, 1);
* gst_value_array_append_value (&v2, &v3);
* g_value_unset (&v3);
* [ Repeat for as many double as your input channels - unset and reinit v3 ]
* g_value_init (&v, GST_TYPE_ARRAY);
* gst_value_array_append_value (&v, &v2);
* g_value_unset (&v2);
* [ Repeat for as many v2's as your output channels - unset and reinit v2]
* g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
* g_value_unset (&v);
* ```
*
*/
const AUDIO_CONVERTER_OPT_MIX_MATRIX: string;
/**
* #GstAudioNoiseShapingMethod, The noise shaping method to use
* to mask noise from quantization errors.
* Default is #GST_AUDIO_NOISE_SHAPING_NONE.
*/
const AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD: string;
/**
* #G_TYPE_UINT, The quantization amount. Components will be
* quantized to multiples of this value.
* Default is 1
*/
const AUDIO_CONVERTER_OPT_QUANTIZATION: string;
/**
* #GstAudioResamplerMethod, The resampler method to use when
* changing sample rates.
* Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
*/
const AUDIO_CONVERTER_OPT_RESAMPLER_METHOD: string;
/**
* Default maximum number of errors tolerated before signaling error.
*/
const AUDIO_DECODER_MAX_ERRORS: number;
/**
* The name of the templates for the sink pad.
*/
const AUDIO_DECODER_SINK_NAME: string;
/**
* The name of the templates for the source pad.
*/
const AUDIO_DECODER_SRC_NAME: string;
/**
* Standard number of channels used in consumer audio.
*/
const AUDIO_DEF_CHANNELS: number;
/**
* Standard format used in consumer audio.
*/
const AUDIO_DEF_FORMAT: string;
/**
* Standard sampling rate used in consumer audio.
*/
const AUDIO_DEF_RATE: number;
/**
* the name of the templates for the sink pad
*/
const AUDIO_ENCODER_SINK_NAME: string;
/**
* the name of the templates for the source pad
*/
const AUDIO_ENCODER_SRC_NAME: string;
/**
* List of all audio formats, for use in template caps strings.
*
* Formats are sorted by decreasing "quality", using these criteria by priority:
* - depth
* - width
* - Float > Signed > Unsigned
* - native endianness preferred
*/
const AUDIO_FORMATS_ALL: string;
/**
* Number of audio formats in #GstAudioFormat.
*/
const AUDIO_FORMAT_LAST: number;
/**
* Maximum range of allowed sample rates, for use in template caps strings.
*/
const AUDIO_RATE_RANGE: string;
/**
* G_TYPE_DOUBLE, B parameter of the cubic filter.
* Values between 0.0 and 2.0 are accepted. 1.0 is the default.
*
* Below are some values of popular filters:
* B C
* Hermite 0.0 0.0
* Spline 1.0 0.0
* Catmull-Rom 0.0 1/2
*/
const AUDIO_RESAMPLER_OPT_CUBIC_B: string;
/**
* G_TYPE_DOUBLE, C parameter of the cubic filter.
* Values between 0.0 and 2.0 are accepted. 0.0 is the default.
*
* See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
*/
const AUDIO_RESAMPLER_OPT_CUBIC_C: string;
/**
* G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
*/
const AUDIO_RESAMPLER_OPT_CUTOFF: string;
/**
* GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be
* interpolated.
* GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.
*/
const AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION: string;
/**
* GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
* constructed.
* GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
*/
const AUDIO_RESAMPLER_OPT_FILTER_MODE: string;
/**
* G_TYPE_UINT: the amount of memory to use for full filter tables before
* switching to interpolated filter tables.
* 1048576 is the default.
*/
const AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD: string;
/**
* G_TYPE_UINT, oversampling to use when interpolating filters
* 8 is the default.
*/
const AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE: string;
/**
* G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
* rates.
* 0.1 is the default.
*/
const AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR: string;
/**
* G_TYPE_INT: the number of taps to use for the filter.
* 0 is the default and selects the taps automatically.
*/
const AUDIO_RESAMPLER_OPT_N_TAPS: string;
/**
* G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
* after the stopband for the kaiser window. 85 dB is the default.
*/
const AUDIO_RESAMPLER_OPT_STOP_ATTENUATION: string;
/**
* G_TYPE_DOUBLE, transition bandwidth. The width of the
* transition band for the kaiser window. 0.087 is the default.
*/
const AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH: string;
const AUDIO_RESAMPLER_QUALITY_DEFAULT: number;
const AUDIO_RESAMPLER_QUALITY_MAX: number;
const AUDIO_RESAMPLER_QUALITY_MIN: number;
/**
* List of all DSD formats, for use in template caps strings.
*
* Big endian formats are preferred, since little-endian ones flip around
* the DSD bytes, and most DSD hardware uses big endian formats.
*/
const DSD_FORMATS_ALL: string;
/**
* The GStreamer media type for DSD.
*/
const DSD_MEDIA_TYPE: string;
/**
* Silence pattern for DSD data.
*
* In DSD, a nullbyte does not correspond to silence. To fill memory regions
* with "DSD silence", these regions must be filled with byte 0x69 instead
* (this is the DSD silence pattern). This constant provides that pattern
* in a more readable fashion.
*/
const DSD_SILENCE_PATTERN_BYTE: number;
/**
* This metadata stays relevant as long as channels are unchanged.
*/
const META_TAG_AUDIO_CHANNELS_STR: string;
/**
* This metadata stays relevant as long as sample rate is unchanged.
*/
const META_TAG_AUDIO_RATE_STR: string;
/**
* This metadata is relevant for audio streams.
*/
const META_TAG_AUDIO_STR: string;
/**
* This metadata stays relevant as long as the DSD plane offsets are unchanged.
*/
const META_TAG_DSD_PLANE_OFFSETS_STR: string;
/**
* Clip the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* `buffer` anymore.
* @param buffer The buffer to clip.
* @param segment Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
* @param rate sample rate.
* @param bpf size of one audio frame in bytes. This is the size of one sample * number of channels.
* @returns %NULL if the buffer is completely outside the configured segment, otherwise the clipped buffer is returned. If the buffer has no timestamp, it is assumed to be inside the segment and is not clipped
*/
function audio_buffer_clip(
buffer: Gst.Buffer,
segment: Gst.Segment,
rate: number,
bpf: number,
): Gst.Buffer | null;
/**
* Maps an audio `gstbuffer` so that it can be read or written and stores the
* result of the map operation in `buffer`.
*
* This is especially useful when the `gstbuffer` is in non-interleaved (planar)
* layout, in which case this function will use the information in the
* `gstbuffer'`s attached #GstAudioMeta in order to map each channel in a
* separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached
* on the `gstbuffer,` then it must be in interleaved layout.
*
* If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked
* against `info`. Normally, they should be equal, but in case they are not,
* a g_critical will be printed and the #GstAudioInfo from the meta will be
* used.
*
* In non-interleaved buffers, it is possible to have each channel on a separate
* #GstMemory. In this case, each memory will be mapped separately to avoid
* copying their contents in a larger memory area. Do note though that it is
* not supported to have a single channel spanning over two or more different
* #GstMemory objects. Although the map operation will likely succeed in this
* case, it will be highly sub-optimal and it is recommended to merge all the
* memories in the buffer before calling this function.
*
* Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid
* as long as it's mapped.
* @param info the audio properties of the buffer
* @param gstbuffer the #GstBuffer to be mapped
* @param flags the access mode for the memory
* @returns %TRUE if the map operation succeeded or %FALSE on failure
*/
function audio_buffer_map(
info: AudioInfo,
gstbuffer: Gst.Buffer,
flags: Gst.MapFlags | null,
): [boolean, AudioBuffer];
/**
* Reorders `buffer` from the channel positions `from` to the channel
* positions `to`. `from` and `to` must contain the same number of
* positions and the same positions, only in a different order.
* `buffer` must be writable.
* @param buffer The buffer to reorder.
* @param format The %GstAudioFormat of the buffer.
* @param from The channel positions in the buffer.
* @param to The channel positions to convert to.
* @returns %TRUE if the reordering was possible.
*/
function audio_buffer_reorder_channels(
buffer: Gst.Buffer,
format: AudioFormat | null,
from: AudioChannelPosition[] | null,
to: AudioChannelPosition[] | null,
): boolean;
/**
* Truncate the buffer to finally have `samples` number of samples, removing
* the necessary amount of samples from the end and `trim` number of samples
* from the beginning.
*
* This function does not know the audio rate, therefore the caller is
* responsible for re-setting the correct timestamp and duration to the
* buffer. However, timestamp will be preserved if trim == 0, and duration
* will also be preserved if there is no trimming to be done. Offset and
* offset end will be preserved / updated.
*
* After calling this function the caller does not own a reference to
* `buffer` anymore.
* @param buffer The buffer to truncate.
* @param bpf size of one audio frame in bytes. This is the size of one sample * number of channels.
* @param trim the number of samples to remove from the beginning of the buffer
* @param samples the final number of samples that should exist in this buffer or -1 to use all the remaining samples if you are only removing samples from the beginning.
* @returns the truncated buffer
*/
function audio_buffer_truncate(buffer: Gst.Buffer, bpf: number, trim: number, samples: number): Gst.Buffer;
/**
* Get the fallback channel-mask for the given number of channels.
*
* This function returns a reasonable fallback channel-mask and should be
* called as a last resort when the specific channel map is unknown.
* @param channels the number of channels
* @returns a fallback channel-mask for @channels or 0 when there is no mask and mono.
*/
function audio_channel_get_fallback_mask(channels: number): number;
/**
* Convert the `channels` present in `channel_mask` to a `position` array
* (which should have at least `channels` entries ensured by caller).
* If `channel_mask` is set to 0, it is considered as 'not present' for purpose
* of conversion.
* A partially valid `channel_mask` with less bits set than the number
* of channels is considered valid.
* @param channel_mask The input channel_mask
* @param position The %GstAudioChannelPosition<!-- -->s
* @returns %TRUE if channel and channel mask are valid and could be converted
*/
function audio_channel_positions_from_mask(
channel_mask: number,
position: AudioChannelPosition[] | null,
): boolean;
/**
* Convert the `position` array of `channels` channels to a bitmask.
*
* If `force_order` is %TRUE it additionally checks if the channels are
* in the order required by GStreamer.
* @param position The %GstAudioChannelPositions
* @param force_order Only consider the GStreamer channel order.
* @returns %TRUE if the channel positions are valid and could be converted.
*/
function audio_channel_positions_to_mask(
position: AudioChannelPosition[] | null,
force_order: boolean,
): [boolean, number];
/**
* Converts `position` to a human-readable string representation for
* debugging purposes.
* @param position The %GstAudioChannelPositions to convert.
* @returns a newly allocated string representing @position
*/
function audio_channel_positions_to_string(position: AudioChannelPosition[] | null): string;
/**
* Reorders the channel positions in `position` from any order to
* the GStreamer channel order.
* @param position The channel positions to reorder to.
* @returns %TRUE if the channel positions are valid and reordering was successful.
*/
function audio_channel_positions_to_valid_order(position: AudioChannelPosition[] | null): boolean;
/**
* Checks if `position` contains valid channel positions for
* `channels` channels. If `force_order` is %TRUE it additionally
* checks if the channels are in the order required by GStreamer.
* @param position The %GstAudioChannelPositions to check.
* @param force_order Only consider the GStreamer channel order.
* @returns %TRUE if the channel positions are valid.
*/
function audio_check_valid_channel_positions(
position: AudioChannelPosition[] | null,
force_order: boolean,
): boolean;
function audio_clipping_meta_api_get_type(): GObject.GType;
function audio_clipping_meta_get_info(): Gst.MetaInfo;
function audio_downmix_meta_api_get_type(): GObject.GType;
function audio_downmix_meta_get_info(): Gst.MetaInfo;
/**
* Construct a #GstAudioFormat with given parameters.
* @param sign signed or unsigned format
* @param endianness G_LITTLE_ENDIAN or G_BIG_ENDIAN
* @param width amount of bits used per sample
* @param depth amount of used bits in @width
* @returns a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format exists with the given parameters.
*/
function audio_format_build_integer(
sign: boolean,
endianness: number,
width: number,
depth: number,
): AudioFormat;
/**
* Fill `length` bytes in `dest` with silence samples for `info`.
* @param info a #GstAudioFormatInfo
* @param dest a destination to fill
*/
function audio_format_fill_silence(info: AudioFormatInfo, dest: Uint8Array | string): void;
/**
* Convert the `format` string to its #GstAudioFormat.
* @param format a format string
* @returns the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the string is not a known format.
*/
function audio_format_from_string(format: string): AudioFormat;
/**
* Get the #GstAudioFormatInfo for `format`
* @param format a #GstAudioFormat
* @returns The #GstAudioFormatInfo for @format.
*/
function audio_format_get_info(format: AudioFormat | null): AudioFormatInfo;
/**
* Returns a string containing a descriptive name for the #GstAudioFormat.
*
* Since 1.26 this can also be used with %GST_AUDIO_FORMAT_UNKNOWN, previous
* versions were printing a critical warning and returned %NULL.
* @param format a #GstAudioFormat audio format
* @returns the name corresponding to @format
*/
function audio_format_to_string(format: AudioFormat | null): string;
/**
* Return all the raw audio formats supported by GStreamer.
* @returns an array of #GstAudioFormat
*/
function audio_formats_raw(): AudioFormat[];
/**
* Returns a reorder map for `from` to `to` that can be used in
* custom channel reordering code, e.g. to convert from or to the
* GStreamer channel order. `from` and `to` must contain the same
* number of positions and the same positions, only in a
* different order.
*
* The resulting `reorder_map` can be used for reordering by assigning
* channel i of the input to channel reorder_map[i] of the output.
* @param from The channel positions to reorder from.
* @param to The channel positions to reorder to.
* @param reorder_map Pointer to the reorder map.
* @returns %TRUE if the channel positions are valid and reordering is possible.
*/
function audio_get_channel_reorder_map(
from: AudioChannelPosition[] | null,
to: AudioChannelPosition[] | null,
reorder_map: number[],
): boolean;
/**
* Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
* payloading type from `spec`.
* @param spec the ringbufer spec
* @returns the size or 0 if the given @type is not supported or cannot be payloaded.
*/
function audio_iec61937_frame_size(spec: AudioRingBufferSpec): number;
/**
* Payloads `src` in the form specified by IEC 61937 for the type from `spec` and
* stores the result in `dst`. `src` must contain exactly one frame of data and
* the frame is not checked for errors.
* @param src a buffer containing the data to payload
* @param dst the destination buffer to store the payloaded contents in. Should not overlap with @src
* @param spec the ringbufer spec for @src
* @param endianness the expected byte order of the payloaded data
* @returns transfer-full: %TRUE if the payloading was successful, %FALSE otherwise.
*/
function audio_iec61937_payload(
src: Uint8Array | string,
dst: Uint8Array | string,
spec: AudioRingBufferSpec,
endianness: number,
): boolean;
/**
* Parse `caps` and update `info`.
* @param caps a #GstCaps
* @returns TRUE if @caps could be parsed
*/
function audio_info_from_caps(caps: Gst.Caps): [boolean, AudioInfo];
/**
* Initialize `info` with default values.
*/
function audio_info_init(): AudioInfo;
/**
* Return the #GType associated with #GstAudioLevelMeta.
* @returns a #GType
*/
function audio_level_meta_api_get_type(): GObject.GType;
/**
* Return the #GstMetaInfo associated with #GstAudioLevelMeta.
* @returns a #GstMetaInfo
*/
function audio_level_meta_get_info(): Gst.MetaInfo;
/**
* Return a generic raw audio caps for formats defined in `formats`.
* If `formats` is %NULL returns a caps for all the supported raw audio formats,
* see gst_audio_formats_raw().
* @param formats an array of raw #GstAudioFormat, or %NULL
* @param layout the layout of audio samples
* @returns an audio @GstCaps
*/
function audio_make_raw_caps(formats: AudioFormat[] | null, layout: AudioLayout | null): Gst.Caps;
function audio_meta_api_get_type(): GObject.GType;
function audio_meta_get_info(): Gst.MetaInfo;
/**
* Reorders `data` from the channel positions `from` to the channel
* positions `to`. `from` and `to` must contain the same number of
* positions and the same positions, only in a different order.
*
* This function internally calls gst_audio_get_channel_reorder_map() and
* gst_audio_reorder_channels_with_reorder_map(). It is more efficient to call
* gst_audio_get_channel_reorder_map() once to retrieve the reorder map and
* then call gst_audio_reorder_channels_with_reorder_map() with the same
* reorder map until the channel positions change.
*
* Note: this function assumes the audio data is in interleaved layout
* @param data The pointer to the memory.
* @param format The %GstAudioFormat of the buffer.
* @param from The channel positions in the buffer.
* @param to The channel positions to convert to.
* @returns %TRUE if the reordering was possible.
*/
function audio_reorder_channels(
data: Uint8Array | string,
format: AudioFormat | null,
from: AudioChannelPosition[] | null,
to: AudioChannelPosition[] | null,
): boolean;
/**
* Reorders `data` with the given `reorder_map`.
*
* The reorder map can be retrieved for example with
* gst_audio_get_channel_reorder_map().
*
* Note: this function assumes the audio data is in interleaved layout
* @param data The pointer to the memory.
* @param bps The number of bytes per sample.
* @param reorder_map The channel reorder map.
*/
function audio_reorder_channels_with_reorder_map(
data: Uint8Array | string,
bps: number,
reorder_map: number[],
): void;
/**
* Make a new resampler.
* @param method a #GstAudioResamplerMethod
* @param flags #GstAudioResamplerFlags
* @param format the #GstAudioFormat
* @param channels the number of channels
* @param in_rate input rate
* @param out_rate output rate
* @param options extra options
* @returns The new #GstAudioResampler.
*/
function audio_resampler_new(
method: AudioResamplerMethod | null,
flags: AudioResamplerFlags | null,
format: AudioFormat | null,
channels: number,
in_rate: number,
out_rate: number,
options: Gst.Structure,
): AudioResampler;
/**
* Set the parameters for resampling from `in_rate` to `out_rate` using `method`
* for `quality` in `options`.
* @param method a #GstAudioResamplerMethod
* @param quality the quality
* @param in_rate the input rate
* @param out_rate the output rate
* @param options a #GstStructure
*/
function audio_resampler_options_set_quality(
method: AudioResamplerMethod | null,
quality: number,
in_rate: number,
out_rate: number,
options: Gst.Structure,
): void;
/**
* Attaches #GstAudioClippingMeta metadata to `buffer` with the given parameters.
* @param buffer a #GstBuffer
* @param format GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
* @param start Amount of audio to clip from start of buffer
* @param end Amount of to clip from end of buffer
* @returns the #GstAudioClippingMeta on @buffer.
*/
function buffer_add_audio_clipping_meta(
buffer: Gst.Buffer,
format: Gst.Format | null,
start: number,
end: number,
): AudioClippingMeta;
/**
* Attaches #GstAudioDownmixMeta metadata to `buffer` with the given parameters.
*
* `matrix` is an two-dimensional array of `to_channels` times `from_channels`
* coefficients, i.e. the i-th output channels is constructed by multiplicating
* the input channels with the coefficients in `matrix[`i] and taking the sum
* of the results.
* @param buffer a #GstBuffer
* @param from_position the channel positions of the source
* @param to_position the channel positions of the destination
* @param matrix The matrix coefficients.
* @returns the #GstAudioDownmixMeta on @buffer.
*/
function buffer_add_audio_downmix_meta(
buffer: Gst.Buffer,
from_position: AudioChannelPosition[] | null,
to_position: AudioChannelPosition[] | null,
matrix: number,
): AudioDownmixMeta;
/**
* Attaches audio level information to `buffer`. (RFC 6464)
* @param buffer a #GstBuffer
* @param level the -dBov from 0-127 (127 is silence).
* @param voice_activity whether the buffer contains voice activity.
* @returns the #GstAudioLevelMeta on @buffer.
*/
function buffer_add_audio_level_meta(
buffer: Gst.Buffer,
level: number,
voice_activity: boolean,
): AudioLevelMeta | null;
/**
* Allocates and attaches a #GstAudioMeta on `buffer,` which must be writable
* for that purpose. The fields of the #GstAudioMeta are directly populated
* from the arguments of this function.
*
* When `info->`layout is %GST_AUDIO_LAYOUT_NON_INTERLEAVED and `offsets` is
* %NULL, the offsets are calculated with a formula that assumes the planes are
* tightly packed and in sequence:
* offsets[channel] = channel * `samples` * sample_stride
*
* It is not allowed for channels to overlap in memory,
* i.e. for each i in [0, channels), the range
* [`offsets[`i], `offsets[`i] + `samples` * sample_stride) must not overlap
* with any other such range. This function will assert if the parameters
* specified cause this restriction to be violated.
*
* It is, obviously, also not allowed to specify parameters that would cause
* out-of-bounds memory access on `buffer`. This is also checked, which means
* that you must add enough memory on the `buffer` before adding this meta.
* @param buffer a #GstBuffer
* @param info the audio properties of the buffer
* @param samples the number of valid samples in the buffer
* @param offsets the offsets (in bytes) where each channel plane starts in the buffer or %NULL to calculate it (see below); must be %NULL also when @info->layout is %GST_AUDIO_LAYOUT_INTERLEAVED
* @returns the #GstAudioMeta that was attached on the @buffer
*/
function buffer_add_audio_meta(
buffer: Gst.Buffer,
info: AudioInfo,
samples: number,
offsets?: number | null,
): AudioMeta;
/**
* Allocates and attaches a #GstDsdPlaneOffsetMeta on `buffer,` which must be
* writable for that purpose. The fields of the #GstDsdPlaneOffsetMeta are
* directly populated from the arguments of this function.
*
* If `offsets` is NULL, then the meta's offsets field is left uninitialized.
* This is useful if for example offset values are to be calculated in the
* meta's offsets field in-place. Similarly, `num_bytes_per_channel` can be
* set to 0, but only if `offsets` is NULL. This is useful if the number of
* bytes per channel is known only later.
*
* It is not allowed for channels to overlap in memory,
* i.e. for each i in [0, channels), the range
* [`offsets[`i], `offsets[`i] + `num_bytes_per_channel)` must not overlap
* with any other such range. This function will assert if the parameters
* specified cause this restriction to be violated.
*
* It is, obviously, also not allowed to specify parameters that would cause
* out-of-bounds memory access on `buffer`. This is also checked, which means
* that you must add enough memory on the `buffer` before adding this meta.
*
* This meta is only needed for non-interleaved (= planar) DSD data.
* @param buffer a #GstBuffer
* @param num_channels Number of channels in the DSD data
* @param num_bytes_per_channel Number of bytes per channel
* @param offsets the offsets (in bytes) where each channel plane starts in the buffer
* @returns the #GstDsdPlaneOffsetMeta that was attached on the @buffer
*/
function buffer_add_dsd_plane_offset_meta(
buffer: Gst.Buffer,
num_channels: number,
num_bytes_per_channel: number,
offsets?: number | null,
): DsdPlaneOffsetMeta;
/**
* Find the #GstAudioDownmixMeta on `buffer` for the given destination
* channel positions.
* @param buffer a #GstBuffer
* @param to_position the channel positions of the destination
* @returns the #GstAudioDownmixMeta on @buffer.
*/
function buffer_get_audio_downmix_meta_for_channels(
buffer: Gst.Buffer,
to_position: AudioChannelPosition[] | null,
): AudioDownmixMeta;
/**
* Find the #GstAudioLevelMeta on `buffer`.
* @param buffer a #GstBuffer
* @returns the #GstAudioLevelMeta or %NULL when there is no such metadata on @buffer.
*/
function buffer_get_audio_level_meta(buffer: Gst.Buffer): AudioLevelMeta | null;
/**
* Converts DSD data from one layout and grouping format to another.
* `num_bytes` must be an integer multiple of the width of both input
* and output format. For example, if the input format is GST_DSD_FORMAT_U32LE,
* and the output format is GST_DSD_FORMAT_U16BE, then `num_bytes` must
* be an integer multiple of both 4 (U32LE width) and 2 (U16BE width).
*
* `reverse_byte_bits` is necessary if the bit order within the DSD bytes
* needs to be reversed. This is rarely necessary, and is not to be
* confused with the endianness of formats (which determines the ordering
* of *bytes*).
*
* `input_plane_offsets` must not be NULL if `input_layout` is set to
* #GST_AUDIO_LAYOUT_NON_INTERLEAVED. The same applies to `output_plane_offsets`.
* These plane offsets define the starting offset of the planes (there is
* exactly one plane per channel) within `input_data` and `output_data`
* respectively. If GST_AUDIO_LAYOUT_INTERLEAVED is used, the plane offsets
* are ignored.
* @param input_data the DSD format conversion's input source
* @param output_data the DSD format conversion's output destination
* @param input_format DSD format of the input data to convert from
* @param output_format DSD format of the output data to convert to
* @param input_layout Input data layout
* @param output_layout Output data layout
* @param input_plane_offsets Plane offsets for non-interleaved input data
* @param output_plane_offsets Plane offsets for non-interleaved output data
* @param num_dsd_bytes How many bytes with DSD data to convert
* @param num_channels Number of channels (must be at least 1)
* @param reverse_byte_bits If TRUE, reverse the bits in each DSD byte
*/
function dsd_convert(
input_data: number,
output_data: number,
input_format: DsdFormat | null,
output_format: DsdFormat | null,
input_layout: AudioLayout | null,
output_layout: AudioLayout | null,
input_plane_offsets: number,
output_plane_offsets: number,
num_dsd_bytes: number,
num_channels: number,
reverse_byte_bits: boolean,
): void;
/**
* Convert the DSD format string `str` to its #GstDsdFormat.
* @param str a DSD format string
* @returns the #GstDsdFormat for @format or GST_DSD_FORMAT_UNKNOWN when the string is not a known format.
*/
function dsd_format_from_string(str: string): DsdFormat;
function dsd_format_get_width(format: DsdFormat | null): number;
/**
* Returns a string containing a descriptive name for
* the #GstDsdFormat if there is one, or NULL otherwise.
* @param format a #GstDsdFormat
* @returns the name corresponding to @format
*/
function dsd_format_to_string(format: DsdFormat | null): string;
/**
* Parse `caps` and update `info`.
* @param caps a #GstCaps
* @returns TRUE if @caps could be parsed
*/
function dsd_info_from_caps(caps: Gst.Caps): [boolean, DsdInfo];
/**
* Initialize `info` with default values.
*/
function dsd_info_init(): DsdInfo;
function dsd_plane_offset_meta_api_get_type(): GObject.GType;
function dsd_plane_offset_meta_get_info(): Gst.MetaInfo;
function stream_volume_convert_volume(
from: StreamVolumeFormat | null,
to: StreamVolumeFormat | null,
val: number,
): number;
interface AudioBaseSinkCustomSlavingCallback {
(
sink: AudioBaseSink,
etime: Gst.ClockTime,
itime: Gst.ClockTime,
requested_skew: Gst.ClockTimeDiff,
discont_reason: AudioBaseSinkDiscontReason,
): void;
}
interface AudioClockGetTimeFunc {
(clock: Gst.Clock): Gst.ClockTime;
}
interface AudioFormatPack {
(
info: AudioFormatInfo,
flags: AudioPackFlags,
src: Uint8Array | string,
data: Uint8Array | string,
length: number,
): void;
}
interface AudioFormatUnpack {
(
info: AudioFormatInfo,
flags: AudioPackFlags,
dest: Uint8Array | string,
data: Uint8Array | string,
length: number,
): void;
}
interface AudioRingBufferCallback {
(rbuf: AudioRingBuffer, data: Uint8Array | string): void;
}
/**
* Flags passed to gst_audio_channel_mixer_new()
*/
/**
* Flags passed to gst_audio_channel_mixer_new()
*/
export namespace AudioChannelMixerFlags {
export const $gtype: GObject.GType<AudioChannelMixerFlags>;
}
enum AudioChannelMixerFlags {
/**
* no flag
*/
NONE,
/**
* input channels are not interleaved
*/
NON_INTERLEAVED_IN,
/**
* output channels are not interleaved
*/
NON_INTERLEAVED_OUT,
/**
* input channels are explicitly unpositioned
*/
UNPOSITIONED_IN,
/**
* output channels are explicitly unpositioned
*/
UNPOSITIONED_OUT,
}
/**
* Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().
*/
/**
* Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().
*/
export namespace AudioConverterFlags {
export const $gtype: GObject.GType<AudioConverterFlags>;
}
enum AudioConverterFlags {
/**
* no flag
*/
NONE,
/**
* the input sample arrays are writable and can be
* used as temporary storage during conversion.
*/
IN_WRITABLE,
/**
* allow arbitrary rate updates with
* gst_audio_converter_update_config().
*/
VARIABLE_RATE,
}
/**
* Extra audio flags
*/
/**
* Extra audio flags
*/
export namespace AudioFlags {
export const $gtype: GObject.GType<AudioFlags>;
}
enum AudioFlags {
/**
* no valid flag
*/
NONE,
/**
* the position array explicitly
* contains unpositioned channels.
*/
UNPOSITIONED,
}
/**
* The different audio flags that a format info can have.
*/
/**
* The different audio flags that a format info can have.
*/
export namespace AudioFormatFlags {
export const $gtype: GObject.GType<AudioFormatFlags>;
}
enum AudioFormatFlags {
/**
* integer samples
*/
INTEGER,
/**
* float samples
*/
FLOAT,
/**
* signed samples
*/
SIGNED,
/**
* complex layout
*/
COMPLEX,
/**
* the format can be used in
* #GstAudioFormatUnpack and #GstAudioFormatPack functions
*/
UNPACK,
}
/**
* The different flags that can be used when packing and unpacking.
*/
/**
* The different flags that can be used when packing and unpacking.
*/
export namespace AudioPackFlags {
export const $gtype: GObject.GType<AudioPackFlags>;
}
enum AudioPackFlags {
/**
* No flag
*/
NONE,
/**
* When the source has a smaller depth
* than the target format, set the least significant bits of the target
* to 0. This is likely slightly faster but less accurate. When this flag
* is not specified, the most significant bits of the source are duplicated
* in the least significant bits of the destination.
*/
TRUNCATE_RANGE,
}
/**
* Extra flags that can be passed to gst_audio_quantize_new()
*/
/**
* Extra flags that can be passed to gst_audio_quantize_new()
*/
export namespace AudioQuantizeFlags {
export const $gtype: GObject.GType<AudioQuantizeFlags>;
}
enum AudioQuantizeFlags {
/**
* no flags
*/
NONE,
/**
* samples are non-interleaved
*/
NON_INTERLEAVED,
}
/**
* Different resampler flags.
*/
/**
* Different resampler flags.
*/
export namespace AudioResamplerFlags {
export const $gtype: GObject.GType<AudioResamplerFlags>;
}
enum AudioResamplerFlags {
/**
* no flags
*/
NONE,
/**
* input samples are non-interleaved.
* an array of blocks of samples, one for each channel, should be passed to the
* resample function.
*/
NON_INTERLEAVED_IN,
/**
* output samples are non-interleaved.
* an array of blocks of samples, one for each channel, should be passed to the
* resample function.
*/
NON_INTERLEAVED_OUT,
/**
* optimize for dynamic updates of the sample
* rates with gst_audio_resampler_update(). This will select an interpolating filter
* when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.
*/
VARIABLE_RATE,
}
namespace AudioAggregator {
// Constructor properties interface
interface ConstructorProps extends GstBase.Aggregator.ConstructorProps {
alignment_threshold: number;
alignmentThreshold: number;
discont_wait: number;
discontWait: number;
force_live: boolean;
forceLive: boolean;
ignore_inactive_pads: boolean;
ignoreInactivePads: boolean;
output_buffer_duration: number;
outputBufferDuration: number;
output_buffer_duration_fraction: Gst.Fraction;
outputBufferDurationFraction: Gst.Fraction;
}
}
/**
* Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
* their source and sink pads,
* gst_element_class_add_static_pad_template_with_gtype() is a convenient
* helper.
*
* #GstAudioAggregator can perform conversion on the data arriving
* on its sink pads, based on the format expected downstream: in order
* to enable that behaviour, the GType of the sink pads must either be
* a (subclass of) #GstAudioAggregatorConvertPad to use the default
* #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
* implementing #GstAudioAggregatorPadClass.convert_buffer.
*
* To allow for the output caps to change, the mechanism is the same as
* above, with the GType of the source pad.
*
* See #GstAudioMixer for an example.
*
* When conversion is enabled, #GstAudioAggregator will accept
* any type of raw audio caps and perform conversion
* on the data arriving on its sink pads, with whatever downstream
* expects as the target format.
*
* In case downstream caps are not fully fixated, it will use
* the first configured sink pad to finish fixating its source pad
* caps.
*
* A notable exception for now is the sample rate, sink pads must
* have the same sample rate as either the downstream requirement,
* or the first configured pad, or a combination of both (when
* downstream specifies a range or a set of acceptable rates).
*
* The #GstAggregator::samples-selected signal is provided with some
* additional information about the output buffer:
* - "offset" G_TYPE_UINT64 Offset in samples since segment start
* for the position that is next to be filled in the output buffer.
* - "frames" G_TYPE_UINT Number of frames per output buffer.
*
* In addition the gst_aggregator_peek_next_sample() function returns
* additional information in the info #GstStructure of the returned sample:
* - "output-offset" G_TYPE_UINT64 Sample offset in output segment relative to
* the output segment's start where the current position of this input
* buffer would be placed
* - "position" G_TYPE_UINT current position in the input buffer in samples
* - "size" G_TYPE_UINT size of the input buffer in samples
*/
abstract class AudioAggregator extends GstBase.Aggregator {
static $gtype: GObject.GType<AudioAggregator>;
// Properties
get alignment_threshold(): number;
set alignment_threshold(val: number);
get alignmentThreshold(): number;
set alignmentThreshold(val: number);
get discont_wait(): number;
set discont_wait(val: number);
get discontWait(): number;
set discontWait(val: number);
/**
* Causes the element to aggregate on a timeout even when no live source is
* connected to its sinks. See #GstAggregator:min-upstream-latency for a
* companion property: in the vast majority of cases where you plan to plug in
* live sources with a non-zero latency, you should set it to a non-zero value.
*/
get force_live(): boolean;
/**
* Causes the element to aggregate on a timeout even when no live source is
* connected to its sinks. See #GstAggregator:min-upstream-latency for a
* companion property: in the vast majority of cases where you plan to plug in
* live sources with a non-zero latency, you should set it to a non-zero value.
*/
get forceLive(): boolean;
/**
* Don't wait for inactive pads when live. An inactive pad
* is a pad that hasn't yet received a buffer, but that has
* been waited on at least once.
*
* The purpose of this property is to avoid aggregating on
* timeout when new pads are requested in advance of receiving
* data flow, for example the user may decide to connect it later,
* but wants to configure it already.
*/
get ignore_inactive_pads(): boolean;
set ignore_inactive_pads(val: boolean);
/**
* Don't wait for inactive pads when live. An inactive pad
* is a pad that hasn't yet received a buffer, but that has
* been waited on at least once.
*
* The purpose of this property is to avoid aggregating on
* timeout when new pads are requested in advance of receiving
* data flow, for example the user may decide to connect it later,
* but wants to configure it already.
*/
get ignoreInactivePads(): boolean;
set ignoreInactivePads(val: boolean);
get output_buffer_duration(): number;
set output_buffer_duration(val: number);
get outputBufferDuration(): number;
set outputBufferDuration(val: number);
/**
* Output block size in nanoseconds, expressed as a fraction.
*/
get output_buffer_duration_fraction(): Gst.Fraction;
set output_buffer_duration_fraction(val: Gst.Fraction);
/**
* Output block size in nanoseconds, expressed as a fraction.
*/
get outputBufferDurationFraction(): Gst.Fraction;
set outputBufferDurationFraction(val: Gst.Fraction);
// Constructors
constructor(properties?: Partial<AudioAggregator.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Aggregates one input buffer to the output
* buffer. The in_offset and out_offset are in "frames", which is
* the size of a sample times the number of channels. Returns TRUE if
* any non-silence was added to the buffer
* @param pad
* @param inbuf
* @param in_offset
* @param outbuf
* @param out_offset
* @param num_frames
*/
vfunc_aggregate_one_buffer(
pad: AudioAggregatorPad,
inbuf: Gst.Buffer,
in_offset: number,
outbuf: Gst.Buffer,
out_offset: number,
num_frames: number,
): boolean;
/**
* Create a new output buffer contains num_frames frames.
* @param num_frames
*/
vfunc_create_output_buffer(num_frames: number): Gst.Buffer;
// Methods
set_sink_caps(pad: AudioAggregatorPad, caps: Gst.Caps): void;
}
namespace AudioAggregatorConvertPad {
// Constructor properties interface
interface ConstructorProps extends AudioAggregatorPad.ConstructorProps {
converter_config: Gst.Structure;
converterConfig: Gst.Structure;
}
}
/**
* An implementation of GstPad that can be used with #GstAudioAggregator.
*
* See #GstAudioAggregator for more details.
*/
class AudioAggregatorConvertPad extends AudioAggregatorPad {
static $gtype: GObject.GType<AudioAggregatorConvertPad>;
// Properties
get converter_config(): Gst.Structure;
set converter_config(val: Gst.Structure);
get converterConfig(): Gst.Structure;
set converterConfig(val: Gst.Structure);
// Constructors
constructor(properties?: Partial<AudioAggregatorConvertPad.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
}
namespace AudioAggregatorPad {
// Constructor properties interface
interface ConstructorProps extends GstBase.AggregatorPad.ConstructorProps {
qos_messages: boolean;
qosMessages: boolean;
}
}
/**
* The default implementation of GstPad used with #GstAudioAggregator
*/
class AudioAggregatorPad extends GstBase.AggregatorPad {
static $gtype: GObject.GType<AudioAggregatorPad>;
// Properties
/**
* Emit QoS messages when dropping buffers.
*/
get qos_messages(): boolean;
set qos_messages(val: boolean);
/**
* Emit QoS messages when dropping buffers.
*/
get qosMessages(): boolean;
set qosMessages(val: boolean);
// Constructors
constructor(properties?: Partial<AudioAggregatorPad.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Convert a buffer from one format to another.
* @param in_info
* @param out_info
* @param buffer
*/
vfunc_convert_buffer(in_info: AudioInfo, out_info: AudioInfo, buffer: Gst.Buffer): Gst.Buffer;
/**
* Called when either the input or output
* formats have changed.
*/
vfunc_update_conversion_info(): void;
}
namespace AudioBaseSink {
// Constructor properties interface
interface ConstructorProps extends GstBase.BaseSink.ConstructorProps {
alignment_threshold: number;
alignmentThreshold: number;
buffer_time: number;
bufferTime: number;
can_activate_pull: boolean | any;
canActivatePull: boolean;
discont_wait: number;
discontWait: number;
drift_tolerance: number;
driftTolerance: number;
latency_time: number;
latencyTime: number;
provide_clock: boolean | any;
provideClock: boolean;
slave_method: AudioBaseSinkSlaveMethod;
slaveMethod: AudioBaseSinkSlaveMethod;
}
}
/**
* 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.
*/
class AudioBaseSink extends GstBase.BaseSink {
static $gtype: GObject.GType<AudioBaseSink>;
// Properties
get alignment_threshold(): number;
set alignment_threshold(val: number);
get alignmentThreshold(): number;
set alignmentThreshold(val: number);
get buffer_time(): number;
set buffer_time(val: number);
get bufferTime(): number;
set bufferTime(val: number);
// This accessor conflicts with a property or field in a parent class or interface.
can_activate_pull: boolean | any;
get canActivatePull(): boolean;
set canActivatePull(val: boolean);
/**
* A window of time in nanoseconds to wait before creating a discontinuity as
* a result of breaching the drift-tolerance.
*/
get discont_wait(): number;
set discont_wait(val: number);
/**
* A window of time in nanoseconds to wait before creating a discontinuity as
* a result of breaching the drift-tolerance.
*/
get discontWait(): number;
set discontWait(val: number);
/**
* Controls the amount of time in microseconds that clocks are allowed
* to drift before resynchronisation happens.
*/
get drift_tolerance(): number;
set drift_tolerance(val: number);
/**
* Controls the amount of time in microseconds that clocks are allowed
* to drift before resynchronisation happens.
*/
get driftTolerance(): number;
set driftTolerance(val: number);
get latency_time(): number;
set latency_time(val: number);
get latencyTime(): number;
set latencyTime(val: number);
// This accessor conflicts with a property or field in a parent class or interface.
provide_clock: boolean | any;
get provideClock(): boolean;
set provideClock(val: boolean);
get slave_method(): AudioBaseSinkSlaveMethod;
set slave_method(val: AudioBaseSinkSlaveMethod);
get slaveMethod(): AudioBaseSinkSlaveMethod;
set slaveMethod(val: AudioBaseSinkSlaveMethod);
// Fields
element: GstBase.BaseSink;
ringbuffer: AudioRingBuffer;
next_sample: number;
provided_clock: Gst.Clock;
eos_rendering: boolean;
// Constructors
constructor(properties?: Partial<AudioBaseSink.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* 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()).
*/
vfunc_create_ringbuffer(): AudioRingBuffer | null;
/**
* 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.
* @param buffer
*/
vfunc_payload(buffer: Gst.Buffer): Gst.Buffer;
// Methods
/**
* 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()).
* @returns The new ringbuffer of @sink.
*/
create_ringbuffer(): AudioRingBuffer | null;
/**
* Get the current alignment threshold, in nanoseconds, used by `sink`.
* @returns The current alignment threshold used by @sink.
*/
get_alignment_threshold(): Gst.ClockTime;
/**
* Get the current discont wait, in nanoseconds, used by `sink`.
* @returns The current discont wait used by @sink.
*/
get_discont_wait(): Gst.ClockTime;
/**
* Get the current drift tolerance, in microseconds, used by `sink`.
* @returns The current drift tolerance used by @sink.
*/
get_drift_tolerance(): number;
/**
* Queries whether `sink` will provide a clock or not. See also
* gst_audio_base_sink_set_provide_clock.
* @returns %TRUE if @sink will provide a clock.
*/
get_provide_clock(): boolean;
/**
* Get the current slave method used by `sink`.
* @returns The current slave method used by @sink.
*/
get_slave_method(): AudioBaseSinkSlaveMethod;
/**
* 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.
*/
report_device_failure(): void;
/**
* Controls the sink's alignment threshold.
* @param alignment_threshold the new alignment threshold in nanoseconds
*/
set_alignment_threshold(alignment_threshold: Gst.ClockTime): void;
/**
* 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.
* @param callback a #GstAudioBaseSinkCustomSlavingCallback
*/
set_custom_slaving_callback(callback: AudioBaseSinkCustomSlavingCallback): void;
/**
* Controls how long the sink will wait before creating a discontinuity.
* @param discont_wait the new discont wait in nanoseconds
*/
set_discont_wait(discont_wait: Gst.ClockTime): void;
/**
* Controls the sink's drift tolerance.
* @param drift_tolerance the new drift tolerance in microseconds
*/
set_drift_tolerance(drift_tolerance: number): void;
/**
* 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.
* @param provide new state
*/
set_provide_clock(provide: boolean): void;
/**
* Controls how clock slaving will be performed in `sink`.
* @param method the new slave method
*/
set_slave_method(method: AudioBaseSinkSlaveMethod | null): void;
}
namespace AudioBaseSrc {
// Constructor properties interface
interface ConstructorProps extends GstBase.PushSrc.ConstructorProps {
actual_buffer_time: number;
actualBufferTime: number;
actual_latency_time: number;
actualLatencyTime: number;
buffer_time: number;
bufferTime: number;
latency_time: number;
latencyTime: number;
provide_clock: boolean | any;
provideClock: boolean;
slave_method: AudioBaseSrcSlaveMethod;
slaveMethod: AudioBaseSrcSlaveMethod;
}
}
/**
* This is the base class for audio sources. Subclasses need to implement the
* ::create_ringbuffer vmethod. This base class will then take care of
* reading samples from the ringbuffer, synchronisation and flushing.
*/
class AudioBaseSrc extends GstBase.PushSrc {
static $gtype: GObject.GType<AudioBaseSrc>;
// Properties
/**
* Actual configured size of audio buffer in microseconds.
*/
get actual_buffer_time(): number;
/**
* Actual configured size of audio buffer in microseconds.
*/
get actualBufferTime(): number;
/**
* Actual configured audio latency in microseconds.
*/
get actual_latency_time(): number;
/**
* Actual configured audio latency in microseconds.
*/
get actualLatencyTime(): number;
get buffer_time(): number;
set buffer_time(val: number);
get bufferTime(): number;
set bufferTime(val: number);
get latency_time(): number;
set latency_time(val: number);
get latencyTime(): number;
set latencyTime(val: number);
// This accessor conflicts with a property or field in a parent class or interface.
provide_clock: boolean | any;
get provideClock(): boolean;
set provideClock(val: boolean);
get slave_method(): AudioBaseSrcSlaveMethod;
set slave_method(val: AudioBaseSrcSlaveMethod);
get slaveMethod(): AudioBaseSrcSlaveMethod;
set slaveMethod(val: AudioBaseSrcSlaveMethod);
// Fields
element: GstBase.PushSrc;
ringbuffer: AudioRingBuffer;
next_sample: number;
clock: Gst.Clock;
// Constructors
constructor(properties?: Partial<AudioBaseSrc.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Create and return the #GstAudioRingBuffer for `src`. This function will call
* the ::create_ringbuffer vmethod and will set `src` as the parent of the
* returned buffer (see gst_object_set_parent()).
*/
vfunc_create_ringbuffer(): AudioRingBuffer | null;
// Methods
/**
* Create and return the #GstAudioRingBuffer for `src`. This function will call
* the ::create_ringbuffer vmethod and will set `src` as the parent of the
* returned buffer (see gst_object_set_parent()).
* @returns The new ringbuffer of @src.
*/
create_ringbuffer(): AudioRingBuffer | null;
/**
* Queries whether `src` will provide a clock or not. See also
* gst_audio_base_src_set_provide_clock.
* @returns %TRUE if @src will provide a clock.
*/
get_provide_clock(): boolean;
/**
* Get the current slave method used by `src`.
* @returns The current slave method used by @src.
*/
get_slave_method(): AudioBaseSrcSlaveMethod;
/**
* Controls whether `src` will provide a clock or not. If `provide` is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of `src`. If `provide` is %FALSE, gst_element_provide_clock() will return NULL.
* @param provide new state
*/
set_provide_clock(provide: boolean): void;
/**
* Controls how clock slaving will be performed in `src`.
* @param method the new slave method
*/
set_slave_method(method: AudioBaseSrcSlaveMethod | null): void;
}
namespace AudioCdSrc {
// Constructor properties interface
interface ConstructorProps extends GstBase.PushSrc.ConstructorProps, Gst.URIHandler.ConstructorProps {
device: string;
mode: AudioCdSrcMode;
track: number;
}
}
/**
* Provides a base class for CD digital audio (CDDA) sources, which handles
* things like seeking, querying, discid calculation, tags, and buffer
* timestamping.
*
* ## Using GstAudioCdSrc-based elements in applications
*
* GstAudioCdSrc registers two #GstFormat<!-- -->s of its own, namely
* the "track" format and the "sector" format. Applications will usually
* only find the "track" format interesting. You can retrieve that #GstFormat
* for use in seek events or queries with gst_format_get_by_nick("track").
*
* In order to query the number of tracks, for example, an application would
* set the CDDA source element to READY or PAUSED state and then query the
* the number of tracks via gst_element_query_duration() using the track
* format acquired above. Applications can query the currently playing track
* in the same way.
*
* Alternatively, applications may retrieve the currently playing track and
* the total number of tracks from the taglist that will posted on the bus
* whenever the CD is opened or the currently playing track changes. The
* taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags.
*
* Applications playing back CD audio using playbin and cdda://n URIs should
* issue a seek command in track format to change between tracks, rather than
* setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin
* involves closing and re-opening the CD device, which is much much slower).
*
* ## Tags and meta-information
*
* CDDA sources will automatically emit a number of tags, details about which
* can be found in the libgsttag documentation. Those tags are:
* #GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL,
* #GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL,
* among others.
*
* ## Tracks and Table of Contents (TOC)
*
* Applications will be informed of the available tracks via a TOC message
* on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for
* each track, with information about each track. The duration for each
* track can be retrieved via the #GST_TAG_DURATION tag from each entry's
* tag list, or calculated via gst_toc_entry_get_start_stop_times().
* The track entries in the TOC will be sorted by track number.
*/
class AudioCdSrc extends GstBase.PushSrc implements Gst.URIHandler {
static $gtype: GObject.GType<AudioCdSrc>;
// Properties
get device(): string;
set device(val: string);
get mode(): AudioCdSrcMode;
set mode(val: AudioCdSrcMode);
get track(): number;
set track(val: number);
// Fields
pushsrc: GstBase.PushSrc;
// Constructors
constructor(properties?: Partial<AudioCdSrc.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* closing the device
*/
vfunc_close(): void;
/**
* opening the device
* @param device
*/
vfunc_open(device: string): boolean;
/**
* reading a sector
* @param sector
*/
vfunc_read_sector(sector: number): Gst.Buffer;
// Methods
/**
* CDDA sources use this function from their start vfunc to announce the
* available data and audio tracks to the base source class. The caller
* should allocate `track` on the stack, the base source will do a shallow
* copy of the structure (and take ownership of the taglist if there is one).
* @param track address of #GstAudioCdSrcTrack to add
* @returns FALSE on error, otherwise TRUE.
*/
add_track(track: AudioCdSrcTrack): boolean;
// Inherited methods
/**
* Gets the list of protocols supported by `handler`. This list may not be
* modified.
* @returns the supported protocols. Returns %NULL if the @handler isn't implemented properly, or the @handler doesn't support any protocols.
*/
get_protocols(): string[] | null;
/**
* Gets the currently handled URI.
* @returns the URI currently handled by the @handler. Returns %NULL if there are no URI currently handled. The returned string must be freed with g_free() when no longer needed.
*/
get_uri(): string | null;
/**
* Gets the type of the given URI handler
* @returns the #GstURIType of the URI handler. Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly.
*/
get_uri_type(): Gst.URIType;
/**
* Tries to set the URI of the given handler.
* @param uri URI to set
* @returns %TRUE if the URI was set successfully, else %FALSE.
*/
set_uri(uri: string): boolean;
/**
* Gets the currently handled URI.
*/
vfunc_get_uri(): string | null;
/**
* Tries to set the URI of the given handler.
* @param uri URI to set
*/
vfunc_set_uri(uri: string): boolean;
/**
* Creates a binding between `source_property` on `source` and `target_property`
* on `target`.
*
* Whenever the `source_property` is changed the `target_property` is
* updated using the same value. For instance:
*
*
* ```c
* g_object_bind_property (action, "active", widget, "sensitive", 0);
* ```
*
*
* Will result in the "sensitive" property of the widget #GObject instance to be
* updated with the same value of the "active" property of the action #GObject
* instance.
*
* If `flags` contains %G_BINDING_BIDIRECTIONAL then the binding will be mutual:
* if `target_property` on `target` changes then the `source_property` on `source`
* will be updated as well.
*
* The binding will automatically be removed when either the `source` or the
* `target` instances are finalized. To remove the binding without affecting the
* `source` and the `target` you can just call g_object_unref() on the returned
* #GBinding instance.
*
* Removing the binding by calling g_object_unref() on it must only be done if
* the binding, `source` and `target` are only used from a single thread and it
* is clear that both `source` and `target` outlive the binding. Especially it
* is not safe to rely on this if the binding, `source` or `target` can be
* finalized from different threads. Keep another reference to the binding and
* use g_binding_unbind() instead to be on the safe side.
*
* A #GObject can have multiple bindings.
* @param source_property the property on @source to bind
* @param target the target #GObject
* @param target_property the property on @target to bind
* @param flags flags to pass to #GBinding
* @returns the #GBinding instance representing the binding between the two #GObject instances. The binding is released whenever the #GBinding reference count reaches zero.
*/
bind_property(
source_property: string,
target: GObject.Object,
target_property: string,
flags: GObject.BindingFlags | null,
): GObject.Binding;
/**
* Complete version of g_object_bind_property().
*
* Creates a binding between `source_property` on `source` and `target_property`
* on `target,` allowing you to set the transformation functions to be used by
* the binding.
*
* If `flags` contains %G_BINDING_BIDIRECTIONAL then the binding will be mutual:
* if `target_property` on `target` changes then the `source_property` on `source`
* will be updated as well. The `transform_from` function is only used in case
* of bidirectional bindings, otherwise it will be ignored
*
* The binding will automatically be removed when either the `source` or the
* `target` instances are finalized. This will release the reference that is
* being held on the #GBinding instance; if you want to hold on to the
* #GBinding instance, you will need to hold a reference to it.
*
* To remove the binding, call g_binding_unbind().
*
* A #GObject can have multiple bindings.
*
* The same `user_data` parameter will be used for both `transform_to`
* and `transform_from` transformation functions; the `notify` function will
* be called once, when the binding is removed. If you need different data
* for each transformation function, please use
* g_object_bind_property_with_closures() instead.
* @param source_property the property on @source to bind
* @param target the target #GObject
* @param target_property the property on @target to bind
* @param flags flags to pass to #GBinding
* @param transform_to the transformation function from the @source to the @target, or %NULL to use the default
* @param transform_from the transformation function from the @target to the @source, or %NULL to use the default
* @param notify a function to call when disposing the binding, to free resources used by the transformation functions, or %NULL if not required
* @returns the #GBinding instance representing the binding between the two #GObject instances. The binding is released whenever the #GBinding reference count reaches zero.
*/
bind_property_full(
source_property: string,
target: GObject.Object,
target_property: string,
flags: GObject.BindingFlags | null,
transform_to?: GObject.BindingTransformFunc | null,
transform_from?: GObject.BindingTransformFunc | null,
notify?: GLib.DestroyNotify | null,
): GObject.Binding;
// Conflicted with GObject.Object.bind_property_full
bind_property_full(...args: never[]): any;
/**
* This function is intended for #GObject implementations to re-enforce
* a [floating][floating-ref] object reference. Doing this is seldom
* required: all #GInitiallyUnowneds are created with a floating reference
* which usually just needs to be sunken by calling g_object_ref_sink().
*/
force_floating(): void;
/**
* Increases the freeze count on `object`. If the freeze count is
* non-zero, the emission of "notify" signals on `object` is
* stopped. The signals are queued until the freeze count is decreased
* to zero. Duplicate notifications are squashed so that at most one
* #GObject::notify signal is emitted for each property modified while the
* object is frozen.
*
* This is necessary for accessors that modify multiple properties to prevent
* premature notification while the object is still being modified.
*/
freeze_notify(): void;
/**
* Gets a named field from the objects table of associations (see g_object_set_data()).
* @param key name of the key for that association
* @returns the data if found, or %NULL if no such data exists.
*/
get_data(key: string): any | null;
/**
* Gets a property of an object.
*
* The value can be:
* - an empty GObject.Value initialized by G_VALUE_INIT, which will be automatically initialized with the expected type of the property (since GLib 2.60)
* - a GObject.Value initialized with the expected type of the property
* - a GObject.Value initialized with a type to which the expected type of the property can be transformed
*
* In general, a copy is made of the property contents and the caller is responsible for freeing the memory by calling GObject.Value.unset.
*
* Note that GObject.Object.get_property is really intended for language bindings, GObject.Object.get is much more convenient for C programming.
* @param property_name The name of the property to get
* @param value Return location for the property value. Can be an empty GObject.Value initialized by G_VALUE_INIT (auto-initialized with expected type since GLib 2.60), a GObject.Value initialized with the expected property type, or a GObject.Value initialized with a transformable type
*/
get_property(property_name: string, value: GObject.Value | any): any;
/**
* This function gets back user data pointers stored via
* g_object_set_qdata().
* @param quark A #GQuark, naming the user data pointer
* @returns The user data pointer set, or %NULL
*/
get_qdata(quark: GLib.Quark): any | null;
/**
* Gets `n_properties` properties for an `object`.
* Obtained properties will be set to `values`. All properties must be valid.
* Warnings will be emitted and undefined behaviour may result if invalid
* properties are passed in.
* @param names the names of each property to get
* @param values the values of each property to get
*/
getv(names: string[], values: (GObject.Value | any)[]): void;
/**
* Checks whether `object` has a [floating][floating-ref] reference.
* @returns %TRUE if @object has a floating reference
*/
is_floating(): boolean;
/**
* Emits a "notify" signal for the property `property_name` on `object`.
*
* When possible, eg. when signaling a property change from within the class
* that registered the property, you should use g_object_notify_by_pspec()
* instead.
*
* Note that emission of the notify signal may be blocked with
* g_object_freeze_notify(). In this case, the signal emissions are queued
* and will be emitted (in reverse order) when g_object_thaw_notify() is
* called.
* @param property_name the name of a property installed on the class of @object.
*/
notify(property_name: string): void;
/**
* Emits a "notify" signal for the property specified by `pspec` on `object`.
*
* This function omits the property name lookup, hence it is faster than
* g_object_notify().
*
* One way to avoid using g_object_notify() from within the
* class that registered the properties, and using g_object_notify_by_pspec()
* instead, is to store the GParamSpec used with
* g_object_class_install_property() inside a static array, e.g.:
*
*
* ```c
* typedef enum
* {
* PROP_FOO = 1,
* PROP_LAST
* } MyObjectProperty;
*
* static GParamSpec *properties[PROP_LAST];
*
* static void
* my_object_class_init (MyObjectClass *klass)
* {
* properties[PROP_FOO] = g_param_spec_int ("foo", NULL, NULL,
* 0, 100,
* 50,
* G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
* g_object_class_install_property (gobject_class,
* PROP_FOO,
* properties[PROP_FOO]);
* }
* ```
*
*
* and then notify a change on the "foo" property with:
*
*
* ```c
* g_object_notify_by_pspec (self, properties[PROP_FOO]);
* ```
*
* @param pspec the #GParamSpec of a property installed on the class of @object.
*/
notify_by_pspec(pspec: GObject.ParamSpec): void;
/**
* Increases the reference count of `object`.
*
* Since GLib 2.56, if `GLIB_VERSION_MAX_ALLOWED` is 2.56 or greater, the type
* of `object` will be propagated to the return type (using the GCC typeof()
* extension), so any casting the caller needs to do on the return type must be
* explicit.
* @returns the same @object
*/
ref(): GObject.Object;
// Conflicted with Gst.Object.ref
ref(...args: never[]): any;
/**
* Increase the reference count of `object,` and possibly remove the
* [floating][floating-ref] reference, if `object` has a floating reference.
*
* In other words, if the object is floating, then this call "assumes
* ownership" of the floating reference, converting it to a normal
* reference by clearing the floating flag while leaving the reference
* count unchanged. If the object is not floating, then this call
* adds a new normal reference increasing the reference count by one.
*
* Since GLib 2.56, the type of `object` will be propagated to the return type
* under the same conditions as for g_object_ref().
* @returns @object
*/
ref_sink(): GObject.Object;
/**
* Releases all references to other objects. This can be used to break
* reference cycles.
*
* This function should only be called from object system implementations.
*/
run_dispose(): void;
/**
* Each object carries around a table of associations from
* strings to pointers. This function lets you set an association.
*
* If the object already had an association with that name,
* the old association will be destroyed.
*
* Internally, the `key` is converted to a #GQuark using g_quark_from_string().
* This means a copy of `key` is kept permanently (even after `object` has been
* finalized) — so it is recommended to only use a small, bounded set of values
* for `key` in your program, to avoid the #GQuark storage growing unbounded.
* @param key name of the key
* @param data data to associate with that key
*/
set_data(key: string, data?: any | null): void;
/**
* Sets a property on an object.
* @param property_name The name of the property to set
* @param value The value to set the property to
*/
set_property(property_name: string, value: GObject.Value | any): void;
/**
* Remove a specified datum from the object's data associations,
* without invoking the association's destroy handler.
* @param key name of the key
* @returns the data if found, or %NULL if no such data exists.
*/
steal_data(key: string): any | null;
/**
* This function gets back user data pointers stored via
* g_object_set_qdata() and removes the `data` from object
* without invoking its destroy() function (if any was
* set).
* Usually, calling this function is only required to update
* user data pointers with a destroy notifier, for example:
*
* ```c
* void
* object_add_to_user_list (GObject *object,
* const gchar *new_string)
* {
* // the quark, naming the object data
* GQuark quark_string_list = g_quark_from_static_string ("my-string-list");
* // retrieve the old string list
* GList *list = g_object_steal_qdata (object, quark_string_list);
*
* // prepend new string
* list = g_list_prepend (list, g_strdup (new_string));
* // this changed 'list', so we need to set it again
* g_object_set_qdata_full (object, quark_string_list, list, free_string_list);
* }
* static void
* free_string_list (gpointer data)
* {
* GList *node, *list = data;
*
* for (node = list; node; node = node->next)
* g_free (node->data);
* g_list_free (list);
* }
* ```
*
* Using g_object_get_qdata() in the above example, instead of
* g_object_steal_qdata() would have left the destroy function set,
* and thus the partial string list would have been freed upon
* g_object_set_qdata_full().
* @param quark A #GQuark, naming the user data pointer
* @returns The user data pointer set, or %NULL
*/
steal_qdata(quark: GLib.Quark): any | null;
/**
* Reverts the effect of a previous call to
* g_object_freeze_notify(). The freeze count is decreased on `object`
* and when it reaches zero, queued "notify" signals are emitted.
*
* Duplicate notifications for each property are squashed so that at most one
* #GObject::notify signal is emitted for each property, in the reverse order
* in which they have been queued.
*
* It is an error to call this function when the freeze count is zero.
*/
thaw_notify(): void;
/**
* Decreases the reference count of `object`. When its reference count
* drops to 0, the object is finalized (i.e. its memory is freed).
*
* If the pointer to the #GObject may be reused in future (for example, if it is
* an instance variable of another object), it is recommended to clear the
* pointer to %NULL rather than retain a dangling pointer to a potentially
* invalid #GObject instance. Use g_clear_object() for this.
*/
unref(): void;
/**
* This function essentially limits the life time of the `closure` to
* the life time of the object. That is, when the object is finalized,
* the `closure` is invalidated by calling g_closure_invalidate() on
* it, in order to prevent invocations of the closure with a finalized
* (nonexisting) object. Also, g_object_ref() and g_object_unref() are
* added as marshal guards to the `closure,` to ensure that an extra
* reference count is held on `object` during invocation of the
* `closure`. Usually, this function will be called on closures that
* use this `object` as closure data.
* @param closure #GClosure to watch
*/
watch_closure(closure: GObject.Closure): void;
/**
* the `constructed` function is called by g_object_new() as the
* final step of the object creation process. At the point of the call, all
* construction properties have been set on the object. The purpose of this
* call is to allow for object initialisation steps that can only be performed
* after construction properties have been set. `constructed` implementors
* should chain up to the `constructed` call of their parent class to allow it
* to complete its initialisation.
*/
vfunc_constructed(): void;
/**
* emits property change notification for a bunch
* of properties. Overriding `dispatch_properties_changed` should be rarely
* needed.
* @param n_pspecs
* @param pspecs
*/
vfunc_dispatch_properties_changed(n_pspecs: number, pspecs: GObject.ParamSpec): void;
/**
* the `dispose` function is supposed to drop all references to other
* objects, but keep the instance otherwise intact, so that client method
* invocations still work. It may be run multiple times (due to reference
* loops). Before returning, `dispose` should chain up to the `dispose` method
* of the parent class.
*/
vfunc_dispose(): void;
/**
* instance finalization function, should finish the finalization of
* the instance begun in `dispose` and chain up to the `finalize` method of the
* parent class.
*/
vfunc_finalize(): void;
/**
* the generic getter for all properties of this type. Should be
* overridden for every type with properties.
* @param property_id
* @param value
* @param pspec
*/
vfunc_get_property(property_id: number, value: GObject.Value | any, pspec: GObject.ParamSpec): void;
/**
* Emits a "notify" signal for the property `property_name` on `object`.
*
* When possible, eg. when signaling a property change from within the class
* that registered the property, you should use g_object_notify_by_pspec()
* instead.
*
* Note that emission of the notify signal may be blocked with
* g_object_freeze_notify(). In this case, the signal emissions are queued
* and will be emitted (in reverse order) when g_object_thaw_notify() is
* called.
* @param pspec
*/
vfunc_notify(pspec: GObject.ParamSpec): void;
/**
* the generic setter for all properties of this type. Should be
* overridden for every type with properties. If implementations of
* `set_property` don't emit property change notification explicitly, this will
* be done implicitly by the type system. However, if the notify signal is
* emitted explicitly, the type system will not emit it a second time.
* @param property_id
* @param value
* @param pspec
*/
vfunc_set_property(property_id: number, value: GObject.Value | any, pspec: GObject.ParamSpec): void;
/**
* Disconnects a handler from an instance so it will not be called during any future or currently ongoing emissions of the signal it has been connected to.
* @param id Handler ID of the handler to be disconnected
*/
disconnect(id: number): void;
/**
* Sets multiple properties of an object at once. The properties argument should be a dictionary mapping property names to values.
* @param properties Object containing the properties to set
*/
set(properties: { [key: string]: any }): void;
/**
* Blocks a handler of an instance so it will not be called during any signal emissions
* @param id Handler ID of the handler to be blocked
*/
block_signal_handler(id: number): void;
/**
* Unblocks a handler so it will be called again during any signal emissions
* @param id Handler ID of the handler to be unblocked
*/
unblock_signal_handler(id: number): void;
/**
* Stops a signal's emission by the given signal name. This will prevent the default handler and any subsequent signal handlers from being invoked.
* @param detailedName Name of the signal to stop emission of
*/
stop_emission_by_name(detailedName: string): void;
}
namespace AudioClock {
// Constructor properties interface
interface ConstructorProps extends Gst.SystemClock.ConstructorProps {}
}
/**
* #GstAudioClock makes it easy for elements to implement a #GstClock, they
* simply need to provide a function that returns the current clock time.
*
* This object is internally used to implement the clock in #GstAudioBaseSink.
*/
class AudioClock extends Gst.SystemClock {
static $gtype: GObject.GType<AudioClock>;
// Fields
clock: Gst.SystemClock;
func: AudioClockGetTimeFunc;
user_data: any;
destroy_notify: GLib.DestroyNotify;
// Constructors
constructor(properties?: Partial<AudioClock.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
static ['new'](name: string, func: AudioClockGetTimeFunc): AudioClock;
// Methods
/**
* Adjust `time` with the internal offset of the audio clock.
* @param time a #GstClockTime
* @returns @time adjusted with the internal offset.
*/
adjust(time: Gst.ClockTime): Gst.ClockTime;
/**
* Report the time as returned by the #GstAudioClockGetTimeFunc without applying
* any offsets.
* @returns the time as reported by the time function of the audio clock
*/
get_time(): Gst.ClockTime;
/**
* Invalidate the clock function. Call this function when the provided
* #GstAudioClockGetTimeFunc cannot be called anymore, for example, when the
* user_data becomes invalid.
*
* After calling this function, `clock` will return the last returned time for
* the rest of its lifetime.
*/
invalidate(): void;
/**
* Inform `clock` that future calls to #GstAudioClockGetTimeFunc will return values
* starting from `time`. The clock will update an internal offset to make sure that
* future calls to internal_time will return an increasing result as required by
* the #GstClock object.
* @param time a #GstClockTime
*/
reset(time: Gst.ClockTime): void;
}
namespace AudioDecoder {
// Constructor properties interface
interface ConstructorProps extends Gst.Element.ConstructorProps {
max_errors: number;
maxErrors: number;
min_latency: number;
minLatency: number;
plc: boolean;
tolerance: number;
}
}
/**
* This base class is for audio decoders turning encoded data into
* raw audio samples.
*
* GstAudioDecoder and subclass should cooperate as follows.
*
* ## Configuration
*
* * Initially, GstAudioDecoder calls `start` when the decoder element
* is activated, which allows subclass to perform any global setup.
* Base class (context) parameters can already be set according to subclass
* capabilities (or possibly upon receive more information in subsequent
* `set_format)`.
* * GstAudioDecoder calls `set_format` to inform subclass of the format
* of input audio data that it is about to receive.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
* * GstAudioDecoder calls `stop` at end of all processing.
*
* As of configuration stage, and throughout processing, GstAudioDecoder
* provides various (context) parameters, e.g. describing the format of
* output audio data (valid when output caps have been set) or current parsing state.
* Conversely, subclass can and should configure context to inform
* base class of its expectation w.r.t. buffer handling.
*
* ## Data processing
* * Base class gathers input data, and optionally allows subclass
* to parse this into subsequently manageable (as defined by subclass)
* chunks. Such chunks are subsequently referred to as 'frames',
* though they may or may not correspond to 1 (or more) audio format frame.
* * Input frame is provided to subclass' `handle_frame`.
* * If codec processing results in decoded data, subclass should call
* `gst_audio_decoder_finish_frame` to have decoded data pushed
* downstream.
* * Just prior to actually pushing a buffer downstream,
* it is passed to `pre_push`. Subclass should either use this callback
* to arrange for additional downstream pushing or otherwise ensure such
* custom pushing occurs after at least a method call has finished since
* setting src pad caps.
* * During the parsing process GstAudioDecoderClass will handle both
* srcpad and sinkpad events. Sink events will be passed to subclass
* if `event` callback has been provided.
*
* ## Shutdown phase
*
* * GstAudioDecoder class calls `stop` to inform the subclass that data
* parsing will be stopped.
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* needs to set the fixed caps on srcpad, when the format is ensured. This
* is typically when base class calls subclass' `set_format` function, though
* it might be delayed until calling `gst_audio_decoder_finish_frame`.
*
* In summary, above process should have subclass concentrating on
* codec data processing while leaving other matters to base class,
* such as most notably timestamp handling. While it may exert more control
* in this area (see e.g. `pre_push)`, it is very much not recommended.
*
* In particular, base class will try to arrange for perfect output timestamps
* as much as possible while tracking upstream timestamps.
* To this end, if deviation between the next ideal expected perfect timestamp
* and upstream exceeds #GstAudioDecoder:tolerance, then resync to upstream
* occurs (which would happen always if the tolerance mechanism is disabled).
*
* In non-live pipelines, baseclass can also (configurably) arrange for
* output buffer aggregation which may help to redue large(r) numbers of
* small(er) buffers being pushed and processed downstream. Note that this
* feature is only available if the buffer layout is interleaved. For planar
* buffers, the decoder implementation is fully responsible for the output
* buffer size.
*
* On the other hand, it should be noted that baseclass only provides limited
* seeking support (upon explicit subclass request), as full-fledged support
* should rather be left to upstream demuxer, parser or alike. This simple
* approach caters for seeking and duration reporting using estimated input
* bitrates.
*
* Things that subclass need to take care of:
*
* * Provide pad templates
* * Set source pad caps when appropriate
* * Set user-configurable properties to sane defaults for format and
* implementing codec at hand, and convey some subclass capabilities and
* expectations in context.
*
* * Accept data in `handle_frame` and provide encoded results to
* `gst_audio_decoder_finish_frame`. If it is prepared to perform
* PLC, it should also accept NULL data in `handle_frame` and provide for
* data for indicated duration.
*/
abstract class AudioDecoder extends Gst.Element {
static $gtype: GObject.GType<AudioDecoder>;
// Properties
/**
* Maximum number of tolerated consecutive decode errors. See
* gst_audio_decoder_set_max_errors() for more details.
*/
get max_errors(): number;
set max_errors(val: number);
/**
* Maximum number of tolerated consecutive decode errors. See
* gst_audio_decoder_set_max_errors() for more details.
*/
get maxErrors(): number;
set maxErrors(val: number);
get min_latency(): number;
set min_latency(val: number);
get minLatency(): number;
set minLatency(val: number);
get plc(): boolean;
set plc(val: boolean);
get tolerance(): number;
set tolerance(val: number);
// Fields
element: Gst.Element;
sinkpad: Gst.Pad;
srcpad: Gst.Pad;
// Constructors
constructor(properties?: Partial<AudioDecoder.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Optional.
* Called when the element changes to GST_STATE_NULL.
* Allows closing external resources.
*/
vfunc_close(): boolean;
/**
* 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.
* @param query
*/
vfunc_decide_allocation(query: Gst.Query): boolean;
/**
* Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned decoded data.
* `hard` indicates whether a FLUSH is being processed,
* or otherwise a DISCONT (or conceptually similar).
* @param hard
*/
vfunc_flush(hard: boolean): void;
/**
* Optional.
* Allows for a custom sink getcaps implementation.
* If not implemented,
* default returns gst_audio_decoder_proxy_getcaps
* applied to sink template caps.
* @param filter
*/
vfunc_getcaps(filter: Gst.Caps): Gst.Caps;
/**
* Provides input data (or NULL to clear any remaining data)
* to subclass. 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 gst_audio_decoder_finish_frame().
* @param buffer
*/
vfunc_handle_frame(buffer: Gst.Buffer): Gst.FlowReturn;
/**
* Negotiate with downstream elements to currently configured #GstAudioInfo.
* Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
* negotiate fails.
*/
vfunc_negotiate(): boolean;
/**
* Optional.
* Called when the element changes to GST_STATE_READY.
* Allows opening external resources.
*/
vfunc_open(): boolean;
vfunc_parse(adapter: GstBase.Adapter): [Gst.FlowReturn, number, number];
/**
* 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.
* @param buffer
*/
vfunc_pre_push(buffer: Gst.Buffer): Gst.FlowReturn;
/**
* Optional.
* Propose buffer allocation parameters for upstream elements.
* Subclasses should chain up to the parent implementation to
* invoke the default handler.
* @param query
*/
vfunc_propose_allocation(query: Gst.Query): boolean;
/**
* Notifies subclass of incoming data format (caps).
* @param caps
*/
vfunc_set_format(caps: Gst.Caps): boolean;
/**
* Optional.
* Event handler on the sink pad. Subclasses should chain up to
* the parent implementation to invoke the default handler.
* @param event
*/
vfunc_sink_event(event: Gst.Event): boolean;
/**
* 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
* @param query
*/
vfunc_sink_query(query: Gst.Query): boolean;
/**
* Optional.
* Event handler on the src pad. Subclasses should chain up to
* the parent implementation to invoke the default handler.
* @param event
*/
vfunc_src_event(event: Gst.Event): boolean;
/**
* 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
* @param query
*/
vfunc_src_query(query: Gst.Query): boolean;
/**
* Optional.
* Called when the element starts processing.
* Allows opening external resources.
*/
vfunc_start(): boolean;
/**
* Optional.
* Called when the element stops processing.
* Allows closing external resources.
*/
vfunc_stop(): boolean;
/**
* 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
* @param outbuf
* @param meta
* @param inbuf
*/
vfunc_transform_meta(outbuf: Gst.Buffer, meta: Gst.Meta, inbuf: Gst.Buffer): boolean;
// Methods
/**
* Helper function that allocates a buffer to hold an audio frame
* for `dec'`s current output format.
* @param size size of the buffer
* @returns allocated buffer
*/
allocate_output_buffer(size: number): Gst.Buffer;
/**
* Collects decoded data and pushes it downstream.
*
* `buf` may be NULL in which case the indicated number of frames
* are discarded and considered to have produced no output
* (e.g. lead-in or setup frames).
* Otherwise, source pad caps must be set when it is called with valid
* data in `buf`.
*
* Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
* invalidated by a call to this function.
* @param buf decoded data
* @param frames number of decoded frames represented by decoded data
* @returns a #GstFlowReturn that should be escalated to caller (of caller)
*/
finish_frame(buf: Gst.Buffer | null, frames: number): Gst.FlowReturn;
/**
* Collects decoded data and pushes it downstream. This function may be called
* multiple times for a given input frame.
*
* `buf` may be NULL in which case it is assumed that the current input frame is
* finished. This is equivalent to calling gst_audio_decoder_finish_subframe()
* with a NULL buffer and frames=1 after having pushed out all decoded audio
* subframes using this function.
*
* When called with valid data in `buf` the source pad caps must have been set
* already.
*
* Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
* invalidated by a call to this function.
* @param buf decoded data
* @returns a #GstFlowReturn that should be escalated to caller (of caller)
*/
finish_subframe(buf?: Gst.Buffer | null): Gst.FlowReturn;
/**
* Lets #GstAudioDecoder sub-classes to know the memory `allocator`
* used by the base class and its `params`.
*
* Unref the `allocator` after use it.
*/
get_allocator(): [Gst.Allocator | null, Gst.AllocationParams | null];
get_audio_info(): AudioInfo;
get_delay(): number;
/**
* Queries decoder drain handling.
* @returns TRUE if drainable handling is enabled. MT safe.
*/
get_drainable(): boolean;
get_estimate_rate(): number;
/**
* Sets the variables pointed to by `min` and `max` to the currently configured
* latency.
*/
get_latency(): [Gst.ClockTime | null, Gst.ClockTime | null];
get_max_errors(): number;
/**
* Queries decoder's latency aggregation.
* @returns aggregation latency. MT safe.
*/
get_min_latency(): Gst.ClockTime;
/**
* Queries decoder required format handling.
* @returns TRUE if required format handling is enabled. MT safe.
*/
get_needs_format(): boolean;
/**
* Return current parsing (sync and eos) state.
*/
get_parse_state(): [boolean, boolean];
/**
* Queries decoder packet loss concealment handling.
* @returns TRUE if packet loss concealment is enabled. MT safe.
*/
get_plc(): boolean;
get_plc_aware(): number;
/**
* Queries current audio jitter tolerance threshold.
* @returns decoder audio jitter tolerance threshold. MT safe.
*/
get_tolerance(): Gst.ClockTime;
/**
* Sets the audio decoder tags and how they should be merged with any
* upstream stream tags. This will override any tags previously-set
* with gst_audio_decoder_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.
* @param tags a #GstTagList to merge, or NULL
* @param mode the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*/
merge_tags(tags: Gst.TagList | null, mode: Gst.TagMergeMode | null): void;
/**
* Negotiate with downstream elements to currently configured #GstAudioInfo.
* Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
* negotiate fails.
* @returns %TRUE if the negotiation succeeded, else %FALSE.
*/
negotiate(): boolean;
/**
* Returns caps that express `caps` (or sink template caps if `caps` == NULL)
* restricted to rate/channels/... combinations supported by downstream
* elements.
* @param caps initial caps
* @param filter filter caps
* @returns a #GstCaps owned by caller
*/
proxy_getcaps(caps?: Gst.Caps | null, filter?: Gst.Caps | null): Gst.Caps;
/**
* Sets a caps in allocation query which are different from the set
* pad's caps. Use this function before calling
* gst_audio_decoder_negotiate(). Setting to %NULL the allocation
* query will use the caps from the pad.
* @param allocation_caps a #GstCaps or %NULL
*/
set_allocation_caps(allocation_caps?: Gst.Caps | null): void;
/**
* Configures decoder drain handling. If drainable, subclass might
* be handed a NULL buffer to have it return any leftover decoded data.
* Otherwise, it is not considered so capable and will only ever be passed
* real data.
*
* MT safe.
* @param enabled new state
*/
set_drainable(enabled: boolean): void;
/**
* Allows baseclass to perform byte to time estimated conversion.
* @param enabled whether to enable byte to time conversion
*/
set_estimate_rate(enabled: boolean): void;
/**
* Sets decoder 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.
* @param min minimum latency
* @param max maximum latency
*/
set_latency(min: Gst.ClockTime, max: Gst.ClockTime): void;
/**
* Sets numbers of tolerated decoder errors, where a tolerated one is then only
* warned about, but more than tolerated will lead to fatal error. You can set
* -1 for never returning fatal errors. Default is set to
* GST_AUDIO_DECODER_MAX_ERRORS.
* @param num max tolerated errors
*/
set_max_errors(num: number): void;
/**
* Sets decoder minimum aggregation latency.
*
* MT safe.
* @param num new minimum latency
*/
set_min_latency(num: Gst.ClockTime): void;
/**
* Configures decoder format needs. If enabled, subclass needs to be
* negotiated with format caps before it can process any data. It will then
* never be handed any data before it has been configured.
* Otherwise, it might be handed data without having been configured and
* is then expected being able to do so either by default
* or based on the input data.
*
* MT safe.
* @param enabled new state
*/
set_needs_format(enabled: boolean): void;
/**
* Configure output caps on the srcpad of `dec`. Similar to
* gst_audio_decoder_set_output_format(), but allows subclasses to specify
* output caps that can't be expressed via #GstAudioInfo e.g. caps that have
* caps features.
* @param caps (fixed) #GstCaps
* @returns %TRUE on success.
*/
set_output_caps(caps: Gst.Caps): boolean;
/**
* Configure output info on the srcpad of `dec`.
* @param info #GstAudioInfo
* @returns %TRUE on success.
*/
set_output_format(info: AudioInfo): boolean;
/**
* Enable or disable decoder packet loss concealment, provided subclass
* and codec are capable and allow handling plc.
*
* MT safe.
* @param enabled new state
*/
set_plc(enabled: boolean): void;
/**
* Indicates whether or not subclass handles packet loss concealment (plc).
* @param plc new plc state
*/
set_plc_aware(plc: boolean): void;
/**
* Configures decoder audio jitter tolerance threshold.
*
* MT safe.
* @param tolerance new tolerance
*/
set_tolerance(tolerance: Gst.ClockTime): void;
/**
* Lets #GstAudioDecoder sub-classes decide if they want the sink pad
* to use the default pad query handler to reply to accept-caps queries.
*
* By setting this to true it is possible to further customize the default
* handler with %GST_PAD_SET_ACCEPT_INTERSECT and
* %GST_PAD_SET_ACCEPT_TEMPLATE
* @param use if the default pad accept-caps query handling should be used
*/
set_use_default_pad_acceptcaps(use: boolean): void;
}
namespace AudioEncoder {
// Constructor properties interface
interface ConstructorProps extends Gst.Element.ConstructorProps, Gst.Preset.ConstructorProps {
hard_resync: boolean;
hardResync: boolean;
mark_granule: boolean;
markGranule: boolean;
perfect_timestamp: boolean;
perfectTimestamp: boolean;
tolerance: number;
}
}
/**
* This base class is for audio encoders turning raw audio samples into
* encoded audio data.
*
* GstAudioEncoder and subclass should cooperate as follows.
*
* ## Configuration
*
* * Initially, GstAudioEncoder calls `start` when the encoder element
* is activated, which allows subclass to perform any global setup.
*
* * GstAudioEncoder calls `set_format` to inform subclass of the format
* of input audio data that it is about to receive. Subclass should
* setup for encoding and configure various base class parameters
* appropriately, notably those directing desired input data handling.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
*
* * GstAudioEncoder calls `stop` at end of all processing.
*
* As of configuration stage, and throughout processing, GstAudioEncoder
* maintains various parameters that provide required context,
* e.g. describing the format of input audio data.
* Conversely, subclass can and should configure these context parameters
* to inform base class of its expectation w.r.t. buffer handling.
*
* ## Data processing
*
* * Base class gathers input sample data (as directed by the context's
* frame_samples and frame_max) and provides this to subclass' `handle_frame`.
* * If codec processing results in encoded data, subclass should call
* gst_audio_encoder_finish_frame() to have encoded data pushed
* downstream. Alternatively, it might also call
* gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
* dropped samples) to indicate dropped (non-encoded) samples.
* * Just prior to actually pushing a buffer downstream,
* it is passed to `pre_push`.
* * During the parsing process GstAudioEncoderClass will handle both
* srcpad and sinkpad events. Sink events will be passed to subclass
* if `event` callback has been provided.
*
* ## Shutdown phase
*
* * GstAudioEncoder class calls `stop` to inform the subclass that data
* parsing will be stopped.
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* needs to set the fixed caps on srcpad, when the format is ensured. This
* is typically when base class calls subclass' `set_format` function, though
* it might be delayed until calling `gst_audio_encoder_finish_frame`.
*
* In summary, above process should have subclass concentrating on
* codec data processing while leaving other matters to base class,
* such as most notably timestamp handling. While it may exert more control
* in this area (see e.g. `pre_push)`, it is very much not recommended.
*
* In particular, base class will either favor tracking upstream timestamps
* (at the possible expense of jitter) or aim to arrange for a perfect stream of
* output timestamps, depending on #GstAudioEncoder:perfect-timestamp.
* However, in the latter case, the input may not be so perfect or ideal, which
* is handled as follows. An input timestamp is compared with the expected
* timestamp as dictated by input sample stream and if the deviation is less
* than #GstAudioEncoder:tolerance, the deviation is discarded.
* Otherwise, it is considered a discontuinity and subsequent output timestamp
* is resynced to the new position after performing configured discontinuity
* processing. In the non-perfect-timestamp case, an upstream variation
* exceeding tolerance only leads to marking DISCONT on subsequent outgoing
* (while timestamps are adjusted to upstream regardless of variation).
* While DISCONT is also marked in the perfect-timestamp case, this one
* optionally (see #GstAudioEncoder:hard-resync)
* performs some additional steps, such as clipping of (early) input samples
* or draining all currently remaining input data, depending on the direction
* of the discontuinity.
*
* If perfect timestamps are arranged, it is also possible to request baseclass
* (usually set by subclass) to provide additional buffer metadata (in OFFSET
* and OFFSET_END) fields according to granule defined semantics currently
* needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
* including buffer) and OFFSET_END to corresponding timestamp (as determined
* by same sample count and sample rate).
*
* Things that subclass need to take care of:
*
* * Provide pad templates
* * Set source pad caps when appropriate
* * Inform base class of buffer processing needs using context's
* frame_samples and frame_bytes.
* * Set user-configurable properties to sane defaults for format and
* implementing codec at hand, e.g. those controlling timestamp behaviour
* and discontinuity processing.
* * Accept data in `handle_frame` and provide encoded results to
* gst_audio_encoder_finish_frame().
*/
abstract class AudioEncoder extends Gst.Element implements Gst.Preset {
static $gtype: GObject.GType<AudioEncoder>;
// Properties
get hard_resync(): boolean;
set hard_resync(val: boolean);
get hardResync(): boolean;
set hardResync(val: boolean);
get mark_granule(): boolean;
get markGranule(): boolean;
get perfect_timestamp(): boolean;
set perfect_timestamp(val: boolean);
get perfectTimestamp(): boolean;
set perfectTimestamp(val: boolean);
get tolerance(): number;
set tolerance(val: number);
// Fields
element: Gst.Element;
sinkpad: Gst.Pad;
srcpad: Gst.Pad;
// Constructors
constructor(properties?: Partial<AudioEncoder.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Optional.
* Called when the element changes to GST_STATE_NULL.
* Allows closing external resources.
*/
vfunc_close(): boolean;
/**
* 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.
* @param query
*/
vfunc_decide_allocation(query: Gst.Query): boolean;
/**
* Optional.
* Instructs subclass to clear any codec caches and discard
* any pending samples and not yet returned encoded data.
*/
vfunc_flush(): void;
/**
* 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.
* @param filter
*/
vfunc_getcaps(filter: Gst.Caps): Gst.Caps;
/**
* 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 gst_audio_encoder_finish_frame().
* @param buffer
*/
vfunc_handle_frame(buffer: Gst.Buffer): Gst.FlowReturn;
/**
* Negotiate with downstream elements to currently configured #GstCaps.
* Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
* negotiate fails.
*/
vfunc_negotiate(): boolean;
/**
* Optional.
* Called when the element changes to GST_STATE_READY.
* Allows opening external resources.
*/
vfunc_open(): boolean;
/**
* 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.
* @param buffer
*/
vfunc_pre_push(buffer: Gst.Buffer): Gst.FlowReturn;
/**
* Optional.
* Propose buffer allocation parameters for upstream elements.
* Subclasses should chain up to the parent implementation to
* invoke the default handler.
* @param query
*/
vfunc_propose_allocation(query: Gst.Query): boolean;
/**
* Notifies subclass of incoming data format.
* GstAudioInfo contains the format according to provided caps.
* @param info
*/
vfunc_set_format(info: AudioInfo): boolean;
/**
* Optional.
* Event handler on the sink pad. Subclasses should chain up to
* the parent implementation to invoke the default handler.
* @param event
*/
vfunc_sink_event(event: Gst.Event): boolean;
/**
* 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
* @param query
*/
vfunc_sink_query(query: Gst.Query): boolean;
/**
* Optional.
* Event handler on the src pad. Subclasses should chain up to
* the parent implementation to invoke the default handler.
* @param event
*/
vfunc_src_event(event: Gst.Event): boolean;
/**
* 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
* @param query
*/
vfunc_src_query(query: Gst.Query): boolean;
/**
* Optional.
* Called when the element starts processing.
* Allows opening external resources.
*/
vfunc_start(): boolean;
/**
* Optional.
* Called when the element stops processing.
* Allows closing external resources.
*/
vfunc_stop(): boolean;
/**
* 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
* @param outbuf
* @param meta
* @param inbuf
*/
vfunc_transform_meta(outbuf: Gst.Buffer, meta: Gst.Meta, inbuf: Gst.Buffer): boolean;
// Methods
/**
* Helper function that allocates a buffer to hold an encoded audio frame
* for `enc'`s current output format.
* @param size size of the buffer
* @returns allocated buffer
*/
allocate_output_buffer(size: number): Gst.Buffer;
/**
* 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 #GstAudioEncoderClass.handle_frame()
* may be invalidated by a call to this function.
* @param buffer encoded data
* @param samples number of samples (per channel) represented by encoded data
* @returns a #GstFlowReturn that should be escalated to caller (of caller)
*/
finish_frame(buffer: Gst.Buffer | null, samples: number): Gst.FlowReturn;
/**
* Lets #GstAudioEncoder sub-classes to know the memory `allocator`
* used by the base class and its `params`.
*
* Unref the `allocator` after use it.
*/
get_allocator(): [Gst.Allocator | null, Gst.AllocationParams | null];
get_audio_info(): AudioInfo;
/**
* Queries encoder drain handling.
* @returns TRUE if drainable handling is enabled. MT safe.
*/
get_drainable(): boolean;
get_frame_max(): number;
get_frame_samples_max(): number;
get_frame_samples_min(): number;
/**
* Queries encoder hard minimum handling.
* @returns TRUE if hard minimum handling is enabled. MT safe.
*/
get_hard_min(): boolean;
get_hard_resync(): boolean;
/**
* Sets the variables pointed to by `min` and `max` to the currently configured
* latency.
*/
get_latency(): [Gst.ClockTime | null, Gst.ClockTime | null];
get_lookahead(): number;
/**
* Queries if the encoder will handle granule marking.
* @returns TRUE if granule marking is enabled. MT safe.
*/
get_mark_granule(): boolean;
/**
* Queries encoder perfect timestamp behaviour.
* @returns TRUE if perfect timestamp setting enabled. MT safe.
*/
get_perfect_timestamp(): boolean;
/**
* Queries current audio jitter tolerance threshold.
* @returns encoder audio jitter tolerance threshold. MT safe.
*/
get_tolerance(): Gst.ClockTime;
/**
* Sets the audio encoder tags and how they should be merged with any
* upstream stream tags. This will override any tags previously-set
* with gst_audio_encoder_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.
* @param tags a #GstTagList to merge, or NULL to unset previously-set tags
* @param mode the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*/
merge_tags(tags: Gst.TagList | null, mode: Gst.TagMergeMode | null): void;
/**
* Negotiate with downstream elements to currently configured #GstCaps.
* Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
* negotiate fails.
* @returns %TRUE if the negotiation succeeded, else %FALSE.
*/
negotiate(): boolean;
/**
* Returns caps that express `caps` (or sink template caps if `caps` == NULL)
* restricted to channel/rate combinations supported by downstream elements
* (e.g. muxers).
* @param caps initial caps
* @param filter filter caps
* @returns a #GstCaps owned by caller
*/
proxy_getcaps(caps?: Gst.Caps | null, filter?: Gst.Caps | null): Gst.Caps;
/**
* Sets a caps in allocation query which are different from the set
* pad's caps. Use this function before calling
* gst_audio_encoder_negotiate(). Setting to %NULL the allocation
* query will use the caps from the pad.
* @param allocation_caps a #GstCaps or %NULL
*/
set_allocation_caps(allocation_caps?: Gst.Caps | null): void;
/**
* 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.
* @param enabled new state
*/
set_drainable(enabled: boolean): void;
/**
* 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
* #GstAudioEncoderClass.set_format() is called.
* @param num number of frames
*/
set_frame_max(num: number): void;
/**
* 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, gst_audio_encoder_set_frame_samples_min()
* must be called with the same number.
*
* Note: This value will be reset to 0 every time before
* #GstAudioEncoderClass.set_format() is called.
* @param num number of samples per frame
*/
set_frame_samples_max(num: number): void;
/**
* 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, gst_audio_encoder_set_frame_samples_max()
* must be called with the same number.
*
* Note: This value will be reset to 0 every time before
* #GstAudioEncoderClass.set_format() is called.
* @param num number of samples per frame
*/
set_frame_samples_min(num: number): void;
/**
* 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.
* @param enabled new state
*/
set_hard_min(enabled: boolean): void;
set_hard_resync(enabled: boolean): void;
/**
* Set the codec headers to be sent downstream whenever requested.
* @param headers a list of #GstBuffer containing the codec header
*/
set_headers(headers: Gst.Buffer[]): void;
/**
* 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.
* @param min minimum latency
* @param max maximum latency
*/
set_latency(min: Gst.ClockTime, max: Gst.ClockTime): void;
/**
* Sets encoder lookahead (in units of input rate samples)
*
* Note: This value will be reset to 0 every time before
* #GstAudioEncoderClass.set_format() is called.
* @param num lookahead
*/
set_lookahead(num: number): void;
/**
* Enable or disable encoder granule handling.
*
* MT safe.
* @param enabled new state
*/
set_mark_granule(enabled: boolean): void;
/**
* Configure output caps on the srcpad of `enc`.
* @param caps #GstCaps
* @returns %TRUE on success.
*/
set_output_format(caps: Gst.Caps): boolean;
/**
* Enable or disable encoder perfect output timestamp preference.
*
* MT safe.
* @param enabled new state
*/
set_perfect_timestamp(enabled: boolean): void;
/**
* Configures encoder audio jitter tolerance threshold.
*
* MT safe.
* @param tolerance new tolerance
*/
set_tolerance(tolerance: Gst.ClockTime): void;
// Inherited methods
/**
* Delete the given preset.
* @param name preset name to remove
* @returns %TRUE for success, %FALSE if e.g. there is no preset with that @name
*/
delete_preset(name: string): boolean;
/**
* Gets the `value` for an existing meta data `tag`. Meta data `tag` names can be
* something like e.g. "comment". Returned values need to be released when done.
* @param name preset name
* @param tag meta data item name
* @returns %TRUE for success, %FALSE if e.g. there is no preset with that @name or no value for the given @tag
*/
get_meta(name: string, tag: string): [boolean, string];
/**
* Get a copy of preset names as a %NULL terminated string array.
* @returns list with names, use g_strfreev() after usage.
*/
get_preset_names(): string[];
/**
* Get a the names of the GObject properties that can be used for presets.
* @returns an array of property names which should be freed with g_strfreev() after use.
*/
get_property_names(): string[];
/**
* Check if one can add new presets, change existing ones and remove presets.
* @returns %TRUE if presets are editable or %FALSE if they are static
*/
is_editable(): boolean;
/**
* Load the given preset.
* @param name preset name to load
* @returns %TRUE for success, %FALSE if e.g. there is no preset with that @name
*/
load_preset(name: string): boolean;
/**
* Renames a preset. If there is already a preset by the `new_name` it will be
* overwritten.
* @param old_name current preset name
* @param new_name new preset name
* @returns %TRUE for success, %FALSE if e.g. there is no preset with @old_name
*/
rename_preset(old_name: string, new_name: string): boolean;
/**
* Save the current object settings as a preset under the given name. If there
* is already a preset by this `name` it will be overwritten.
* @param name preset name to save
* @returns %TRUE for success, %FALSE
*/
save_preset(name: string): boolean;
/**
* Sets a new `value` for an existing meta data item or adds a new item. Meta
* data `tag` names can be something like e.g. "comment". Supplying %NULL for the
* `value` will unset an existing value.
* @param name preset name
* @param tag meta data item name
* @param value new value
* @returns %TRUE for success, %FALSE if e.g. there is no preset with that @name
*/
set_meta(name: string, tag: string, value?: string | null): boolean;
/**
* Delete the given preset.
* @param name preset name to remove
*/
vfunc_delete_preset(name: string): boolean;
/**
* Gets the `value` for an existing meta data `tag`. Meta data `tag` names can be
* something like e.g. "comment". Returned values need to be released when done.
* @param name preset name
* @param tag meta data item name
*/
vfunc_get_meta(name: string, tag: string): [boolean, string];
/**
* Get a copy of preset names as a %NULL terminated string array.
*/
vfunc_get_preset_names(): string[];
/**
* Get a the names of the GObject properties that can be used for presets.
*/
vfunc_get_property_names(): string[];
/**
* Load the given preset.
* @param name preset name to load
*/
vfunc_load_preset(name: string): boolean;
/**
* Renames a preset. If there is already a preset by the `new_name` it will be
* overwritten.
* @param old_name current preset name
* @param new_name new preset name
*/
vfunc_rename_preset(old_name: string, new_name: string): boolean;
/**
* Save the current object settings as a preset under the given name. If there
* is already a preset by this `name` it will be overwritten.
* @param name preset name to save
*/
vfunc_save_preset(name: string): boolean;
/**
* Sets a new `value` for an existing meta data item or adds a new item. Meta
* data `tag` names can be something like e.g. "comment". Supplying %NULL for the
* `value` will unset an existing value.
* @param name preset name
* @param tag meta data item name
* @param value new value
*/
vfunc_set_meta(name: string, tag: string, value?: string | null): boolean;
/**
* Creates a binding between `source_property` on `source` and `target_property`
* on `target`.
*
* Whenever the `source_property` is changed the `target_property` is
* updated using the same value. For instance:
*
*
* ```c
* g_object_bind_property (action, "active", widget, "sensitive", 0);
* ```
*
*
* Will result in the "sensitive" property of the widget #GObject instance to be
* updated with the same value of the "active" property of the action #GObject
* instance.
*
* If `flags` contains %G_BINDING_BIDIRECTIONAL then the binding will be mutual:
* if `target_property` on `target` changes then the `source_property` on `source`
* will be updated as well.
*
* The binding will automatically be removed when either the `source` or the
* `target` instances are finalized. To remove the binding without affecting the
* `source` and the `target` you can just call g_object_unref() on the returned
* #GBinding instance.
*
* Removing the binding by calling g_object_unref() on it must only be done if
* the binding, `source` and `target` are only used from a single thread and it
* is clear that both `source` and `target` outlive the binding. Especially it
* is not safe to rely on this if the binding, `source` or `target` can be
* finalized from different threads. Keep another reference to the binding and
* use g_binding_unbind() instead to be on the safe side.
*
* A #GObject can have multiple bindings.
* @param source_property the property on @source to bind
* @param target the target #GObject
* @param target_property the property on @target to bind
* @param flags flags to pass to #GBinding
* @returns the #GBinding instance representing the binding between the two #GObject instances. The binding is released whenever the #GBinding reference count reaches zero.
*/
bind_property(
source_property: string,
target: GObject.Object,
target_property: string,
flags: GObject.BindingFlags | null,
): GObject.Binding;
/**
* Complete version of g_object_bind_property().
*
* Creates a binding between `source_property` on `source` and `target_property`
* on `target,` allowing you to set the transformation functions to be used by
* the binding.
*
* If `flags` contains %G_BINDING_BIDIRECTIONAL then the binding will be mutual:
* if `target_property` on `target` changes then the `source_property` on `source`
* will be updated as well. The `transform_from` function is only used in case
* of bidirectional bindings, otherwise it will be ignored
*
* The binding will automatically be removed when either the `source` or the
* `target` instances are finalized. This will release the reference that is
* being held on the #GBinding instance; if you want to hold on to the
* #GBinding instance, you will need to hold a reference to it.
*
* To remove the binding, call g_binding_unbind().
*
* A #GObject can have multiple bindings.
*
* The same `user_data` parameter will be used for both `transform_to`
* and `transform_from` transformation functions; the `notify` function will
* be called once, when the binding is removed. If you need different data
* for each transformation function, please use
* g_object_bind_property_with_closures() instead.
* @param source_property the property on @source to bind
* @param target the target #GObject
* @param target_property the property on @target to bind
* @param flags flags to pass to #GBinding
* @param transform_to the transformation function from the @source to the @target, or %NULL to use the default
* @param transform_from the transformation function from the @target to the @source, or %NULL to use the default
* @param notify a function to call when disposing the binding, to free resources used by the transformation functions, or %NULL if not required
* @returns the #GBinding instance representing the binding between the two #GObject instances. The binding is released whenever the #GBinding reference count reaches zero.
*/
bind_property_full(
source_property: string,
target: GObject.Object,
target_property: string,
flags: GObject.BindingFlags | null,
transform_to?: GObject.BindingTransformFunc | null,
transform_from?: GObject.BindingTransformFunc | null,
notify?: GLib.DestroyNotify | null,
): GObject.Binding;
// Conflicted with GObject.Object.bind_property_full
bind_property_full(...args: never[]): any;
/**
* This function is intended for #GObject implementations to re-enforce
* a [floating][floating-ref] object reference. Doing this is seldom
* required: all #GInitiallyUnowneds are created with a floating reference
* which usually just needs to be sunken by calling g_object_ref_sink().
*/
force_floating(): void;
/**
* Increases the freeze count on `object`. If the freeze count is
* non-zero, the emission of "notify" signals on `object` is
* stopped. The signals are queued until the freeze count is decreased
* to zero. Duplicate notifications are squashed so that at most one
* #GObject::notify signal is emitted for each property modified while the
* object is frozen.
*
* This is necessary for accessors that modify multiple properties to prevent
* premature notification while the object is still being modified.
*/
freeze_notify(): void;
/**
* Gets a named field from the objects table of associations (see g_object_set_data()).
* @param key name of the key for that association
* @returns the data if found, or %NULL if no such data exists.
*/
get_data(key: string): any | null;
/**
* Gets a property of an object.
*
* The value can be:
* - an empty GObject.Value initialized by G_VALUE_INIT, which will be automatically initialized with the expected type of the property (since GLib 2.60)
* - a GObject.Value initialized with the expected type of the property
* - a GObject.Value initialized with a type to which the expected type of the property can be transformed
*
* In general, a copy is made of the property contents and the caller is responsible for freeing the memory by calling GObject.Value.unset.
*
* Note that GObject.Object.get_property is really intended for language bindings, GObject.Object.get is much more convenient for C programming.
* @param property_name The name of the property to get
* @param value Return location for the property value. Can be an empty GObject.Value initialized by G_VALUE_INIT (auto-initialized with expected type since GLib 2.60), a GObject.Value initialized with the expected property type, or a GObject.Value initialized with a transformable type
*/
get_property(property_name: string, value: GObject.Value | any): any;
/**
* This function gets back user data pointers stored via
* g_object_set_qdata().
* @param quark A #GQuark, naming the user data pointer
* @returns The user data pointer set, or %NULL
*/
get_qdata(quark: GLib.Quark): any | null;
/**
* Gets `n_properties` properties for an `object`.
* Obtained properties will be set to `values`. All properties must be valid.
* Warnings will be emitted and undefined behaviour may result if invalid
* properties are passed in.
* @param names the names of each property to get
* @param values the values of each property to get
*/
getv(names: string[], values: (GObject.Value | any)[]): void;
/**
* Checks whether `object` has a [floating][floating-ref] reference.
* @returns %TRUE if @object has a floating reference
*/
is_floating(): boolean;
/**
* Emits a "notify" signal for the property `property_name` on `object`.
*
* When possible, eg. when signaling a property change from within the class
* that registered the property, you should use g_object_notify_by_pspec()
* instead.
*
* Note that emission of the notify signal may be blocked with
* g_object_freeze_notify(). In this case, the signal emissions are queued
* and will be emitted (in reverse order) when g_object_thaw_notify() is
* called.
* @param property_name the name of a property installed on the class of @object.
*/
notify(property_name: string): void;
/**
* Emits a "notify" signal for the property specified by `pspec` on `object`.
*
* This function omits the property name lookup, hence it is faster than
* g_object_notify().
*
* One way to avoid using g_object_notify() from within the
* class that registered the properties, and using g_object_notify_by_pspec()
* instead, is to store the GParamSpec used with
* g_object_class_install_property() inside a static array, e.g.:
*
*
* ```c
* typedef enum
* {
* PROP_FOO = 1,
* PROP_LAST
* } MyObjectProperty;
*
* static GParamSpec *properties[PROP_LAST];
*
* static void
* my_object_class_init (MyObjectClass *klass)
* {
* properties[PROP_FOO] = g_param_spec_int ("foo", NULL, NULL,
* 0, 100,
* 50,
* G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
* g_object_class_install_property (gobject_class,
* PROP_FOO,
* properties[PROP_FOO]);
* }
* ```
*
*
* and then notify a change on the "foo" property with:
*
*
* ```c
* g_object_notify_by_pspec (self, properties[PROP_FOO]);
* ```
*
* @param pspec the #GParamSpec of a property installed on the class of @object.
*/
notify_by_pspec(pspec: GObject.ParamSpec): void;
/**
* Increases the reference count of `object`.
*
* Since GLib 2.56, if `GLIB_VERSION_MAX_ALLOWED` is 2.56 or greater, the type
* of `object` will be propagated to the return type (using the GCC typeof()
* extension), so any casting the caller needs to do on the return type must be
* explicit.
* @returns the same @object
*/
ref(): GObject.Object;
// Conflicted with Gst.Object.ref
ref(...args: never[]): any;
/**
* Increase the reference count of `object,` and possibly remove the
* [floating][floating-ref] reference, if `object` has a floating reference.
*
* In other words, if the object is floating, then this call "assumes
* ownership" of the floating reference, converting it to a normal
* reference by clearing the floating flag while leaving the reference
* count unchanged. If the object is not floating, then this call
* adds a new normal reference increasing the reference count by one.
*
* Since GLib 2.56, the type of `object` will be propagated to the return type
* under the same conditions as for g_object_ref().
* @returns @object
*/
ref_sink(): GObject.Object;
/**
* Releases all references to other objects. This can be used to break
* reference cycles.
*
* This function should only be called from object system implementations.
*/
run_dispose(): void;
/**
* Each object carries around a table of associations from
* strings to pointers. This function lets you set an association.
*
* If the object already had an association with that name,
* the old association will be destroyed.
*
* Internally, the `key` is converted to a #GQuark using g_quark_from_string().
* This means a copy of `key` is kept permanently (even after `object` has been
* finalized) — so it is recommended to only use a small, bounded set of values
* for `key` in your program, to avoid the #GQuark storage growing unbounded.
* @param key name of the key
* @param data data to associate with that key
*/
set_data(key: string, data?: any | null): void;
/**
* Sets a property on an object.
* @param property_name The name of the property to set
* @param value The value to set the property to
*/
set_property(property_name: string, value: GObject.Value | any): void;
/**
* Remove a specified datum from the object's data associations,
* without invoking the association's destroy handler.
* @param key name of the key
* @returns the data if found, or %NULL if no such data exists.
*/
steal_data(key: string): any | null;
/**
* This function gets back user data pointers stored via
* g_object_set_qdata() and removes the `data` from object
* without invoking its destroy() function (if any was
* set).
* Usually, calling this function is only required to update
* user data pointers with a destroy notifier, for example:
*
* ```c
* void
* object_add_to_user_list (GObject *object,
* const gchar *new_string)
* {
* // the quark, naming the object data
* GQuark quark_string_list = g_quark_from_static_string ("my-string-list");
* // retrieve the old string list
* GList *list = g_object_steal_qdata (object, quark_string_list);
*
* // prepend new string
* list = g_list_prepend (list, g_strdup (new_string));
* // this changed 'list', so we need to set it again
* g_object_set_qdata_full (object, quark_string_list, list, free_string_list);
* }
* static void
* free_string_list (gpointer data)
* {
* GList *node, *list = data;
*
* for (node = list; node; node = node->next)
* g_free (node->data);
* g_list_free (list);
* }
* ```
*
* Using g_object_get_qdata() in the above example, instead of
* g_object_steal_qdata() would have left the destroy function set,
* and thus the partial string list would have been freed upon
* g_object_set_qdata_full().
* @param quark A #GQuark, naming the user data pointer
* @returns The user data pointer set, or %NULL
*/
steal_qdata(quark: GLib.Quark): any | null;
/**
* Reverts the effect of a previous call to
* g_object_freeze_notify(). The freeze count is decreased on `object`
* and when it reaches zero, queued "notify" signals are emitted.
*
* Duplicate notifications for each property are squashed so that at most one
* #GObject::notify signal is emitted for each property, in the reverse order
* in which they have been queued.
*
* It is an error to call this function when the freeze count is zero.
*/
thaw_notify(): void;
/**
* Decreases the reference count of `object`. When its reference count
* drops to 0, the object is finalized (i.e. its memory is freed).
*
* If the pointer to the #GObject may be reused in future (for example, if it is
* an instance variable of another object), it is recommended to clear the
* pointer to %NULL rather than retain a dangling pointer to a potentially
* invalid #GObject instance. Use g_clear_object() for this.
*/
unref(): void;
/**
* This function essentially limits the life time of the `closure` to
* the life time of the object. That is, when the object is finalized,
* the `closure` is invalidated by calling g_closure_invalidate() on
* it, in order to prevent invocations of the closure with a finalized
* (nonexisting) object. Also, g_object_ref() and g_object_unref() are
* added as marshal guards to the `closure,` to ensure that an extra
* reference count is held on `object` during invocation of the
* `closure`. Usually, this function will be called on closures that
* use this `object` as closure data.
* @param closure #GClosure to watch
*/
watch_closure(closure: GObject.Closure): void;
/**
* the `constructed` function is called by g_object_new() as the
* final step of the object creation process. At the point of the call, all
* construction properties have been set on the object. The purpose of this
* call is to allow for object initialisation steps that can only be performed
* after construction properties have been set. `constructed` implementors
* should chain up to the `constructed` call of their parent class to allow it
* to complete its initialisation.
*/
vfunc_constructed(): void;
/**
* emits property change notification for a bunch
* of properties. Overriding `dispatch_properties_changed` should be rarely
* needed.
* @param n_pspecs
* @param pspecs
*/
vfunc_dispatch_properties_changed(n_pspecs: number, pspecs: GObject.ParamSpec): void;
/**
* the `dispose` function is supposed to drop all references to other
* objects, but keep the instance otherwise intact, so that client method
* invocations still work. It may be run multiple times (due to reference
* loops). Before returning, `dispose` should chain up to the `dispose` method
* of the parent class.
*/
vfunc_dispose(): void;
/**
* instance finalization function, should finish the finalization of
* the instance begun in `dispose` and chain up to the `finalize` method of the
* parent class.
*/
vfunc_finalize(): void;
/**
* the generic getter for all properties of this type. Should be
* overridden for every type with properties.
* @param property_id
* @param value
* @param pspec
*/
vfunc_get_property(property_id: number, value: GObject.Value | any, pspec: GObject.ParamSpec): void;
/**
* Emits a "notify" signal for the property `property_name` on `object`.
*
* When possible, eg. when signaling a property change from within the class
* that registered the property, you should use g_object_notify_by_pspec()
* instead.
*
* Note that emission of the notify signal may be blocked with
* g_object_freeze_notify(). In this case, the signal emissions are queued
* and will be emitted (in reverse order) when g_object_thaw_notify() is
* called.
* @param pspec
*/
vfunc_notify(pspec: GObject.ParamSpec): void;
/**
* the generic setter for all properties of this type. Should be
* overridden for every type with properties. If implementations of
* `set_property` don't emit property change notification explicitly, this will
* be done implicitly by the type system. However, if the notify signal is
* emitted explicitly, the type system will not emit it a second time.
* @param property_id
* @param value
* @param pspec
*/
vfunc_set_property(property_id: number, value: GObject.Value | any, pspec: GObject.ParamSpec): void;
/**
* Disconnects a handler from an instance so it will not be called during any future or currently ongoing emissions of the signal it has been connected to.
* @param id Handler ID of the handler to be disconnected
*/
disconnect(id: number): void;
/**
* Sets multiple properties of an object at once. The properties argument should be a dictionary mapping property names to values.
* @param properties Object containing the properties to set
*/
set(properties: { [key: string]: any }): void;
/**
* Blocks a handler of an instance so it will not be called during any signal emissions
* @param id Handler ID of the handler to be blocked
*/
block_signal_handler(id: number): void;
/**
* Unblocks a handler so it will be called again during any signal emissions
* @param id Handler ID of the handler to be unblocked
*/
unblock_signal_handler(id: number): void;
/**
* Stops a signal's emission by the given signal name. This will prevent the default handler and any subsequent signal handlers from being invoked.
* @param detailedName Name of the signal to stop emission of
*/
stop_emission_by_name(detailedName: string): void;
}
namespace AudioFilter {
// Constructor properties interface
interface ConstructorProps extends GstBase.BaseTransform.ConstructorProps {}
}
/**
* #GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio
* filters, ie. those that output the same format that they get as input.
*
* #GstAudioFilter will parse the input format for you (with error checking)
* before calling your setup function. Also, elements deriving from
* #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
* their class_init function to easily configure the set of caps/formats that
* the element is able to handle.
*
* Derived classes should override the #GstAudioFilterClass.setup() and
* #GstBaseTransformClass.transform_ip() and/or
* #GstBaseTransformClass.transform()
* virtual functions in their class_init function.
*/
abstract class AudioFilter extends GstBase.BaseTransform {
static $gtype: GObject.GType<AudioFilter>;
// Fields
basetransform: GstBase.BaseTransform;
// Constructors
constructor(properties?: Partial<AudioFilter.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Static methods
static add_pad_templates(allowed_caps: Gst.Caps): void;
// Virtual methods
/**
* virtual function called whenever the format changes
* @param info
*/
vfunc_setup(info: AudioInfo): boolean;
}
namespace AudioRingBuffer {
// Constructor properties interface
interface ConstructorProps extends Gst.Object.ConstructorProps {}
}
/**
* This object is the base class for audio ringbuffers used by the base
* audio source and sink classes.
*
* The ringbuffer abstracts a circular buffer of data. One reader and
* one writer can operate on the data from different threads in a lockfree
* manner. The base class is sufficiently flexible to be used as an
* abstraction for DMA based ringbuffers as well as a pure software
* implementations.
*/
abstract class AudioRingBuffer extends Gst.Object {
static $gtype: GObject.GType<AudioRingBuffer>;
// Fields
object: Gst.Object;
open: boolean;
acquired: boolean;
memory: number;
size: number;
samples_per_seg: number;
empty_seg: number;
state: number;
segdone: number;
segbase: number;
waiting: number;
// Constructors
constructor(properties?: Partial<AudioRingBuffer.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Static methods
/**
* Print debug info about the buffer sized in `spec` to the debug log.
* @param spec the spec to debug
*/
static debug_spec_buff(spec: AudioRingBufferSpec): void;
/**
* Print debug info about the parsed caps in `spec` to the debug log.
* @param spec the spec to debug
*/
static debug_spec_caps(spec: AudioRingBufferSpec): void;
/**
* Parse `caps` into `spec`.
* @param spec a spec
* @param caps a #GstCaps
*/
static parse_caps(spec: AudioRingBufferSpec, caps: Gst.Caps): boolean;
// Virtual methods
/**
* Allocate the resources for the ringbuffer. This function fills
* in the data pointer of the ring buffer with a valid #GstBuffer
* to which samples can be written.
* @param spec the specs of the buffer
*/
vfunc_acquire(spec: AudioRingBufferSpec): boolean;
/**
* Activate `buf` to start or stop pulling data.
*
* MT safe.
* @param active the new mode
*/
vfunc_activate(active: boolean): boolean;
/**
* Clear all samples from the ringbuffer.
*
* MT safe.
*/
vfunc_clear_all(): void;
/**
* Close the audio device associated with the ring buffer. The ring buffer
* should already have been released via gst_audio_ring_buffer_release().
*/
vfunc_close_device(): boolean;
/**
* Commit `in_samples` samples pointed to by `data` to the ringbuffer `buf`.
*
* `in_samples` and `out_samples` define the rate conversion to perform on the
* samples in `data`. For negative rates, `out_samples` must be negative and
* `in_samples` positive.
*
* When `out_samples` is positive, the first sample will be written at position `sample`
* in the ringbuffer. When `out_samples` is negative, the last sample will be written to
* `sample` in reverse order.
*
* `out_samples` does not need to be a multiple of the segment size of the ringbuffer
* although it is recommended for optimal performance.
*
* `accum` will hold a temporary accumulator used in rate conversion and should be
* set to 0 when this function is first called. In case the commit operation is
* interrupted, one can resume the processing by passing the previously returned
* `accum` value back to this function.
*
* MT safe.
* @param sample the sample position of the data
* @param data the data to commit
* @param out_samples the number of samples to write to the ringbuffer
* @param accum accumulator for rate conversion.
*/
vfunc_commit(
sample: number,
data: Uint8Array | string,
out_samples: number,
accum: number,
): [number, number, number];
/**
* Get the number of samples queued in the audio device. This is
* usually less than the segment size but can be bigger when the
* implementation uses another internal buffer between the audio
* device.
*
* For playback ringbuffers this is the amount of samples transferred from the
* ringbuffer to the device but still not played.
*
* For capture ringbuffers this is the amount of samples in the device that are
* not yet transferred to the ringbuffer.
*/
vfunc_delay(): number;
/**
* Open the audio device associated with the ring buffer. Does not perform any
* setup on the device. You must open the device before acquiring the ring
* buffer.
*/
vfunc_open_device(): boolean;
/**
* Pause processing samples from the ringbuffer.
*/
vfunc_pause(): boolean;
/**
* Free the resources of the ringbuffer.
*/
vfunc_release(): boolean;
/**
* resume processing of samples after pause
*/
vfunc_resume(): boolean;
/**
* Start processing samples from the ringbuffer.
*/
vfunc_start(): boolean;
/**
* Stop processing samples from the ringbuffer.
*/
vfunc_stop(): boolean;
// Methods
/**
* Allocate the resources for the ringbuffer. This function fills
* in the data pointer of the ring buffer with a valid #GstBuffer
* to which samples can be written.
* @param spec the specs of the buffer
* @returns TRUE if the device could be acquired, FALSE on error. MT safe.
*/
acquire(spec: AudioRingBufferSpec): boolean;
/**
* Activate `buf` to start or stop pulling data.
*
* MT safe.
* @param active the new mode
* @returns TRUE if the device could be activated in the requested mode, FALSE on error.
*/
activate(active: boolean): boolean;
/**
* Subclasses should call this function to notify the fact that
* `advance` segments are now processed by the device.
*
* MT safe.
* @param advance the number of segments written
*/
advance(advance: number): void;
/**
* Clear the given segment of the buffer with silence samples.
* This function is used by subclasses.
*
* MT safe.
* @param segment the segment to clear
*/
clear(segment: number): void;
/**
* Clear all samples from the ringbuffer.
*
* MT safe.
*/
clear_all(): void;
/**
* Close the audio device associated with the ring buffer. The ring buffer
* should already have been released via gst_audio_ring_buffer_release().
* @returns TRUE if the device could be closed, FALSE on error. MT safe.
*/
close_device(): boolean;
/**
* Commit `in_samples` samples pointed to by `data` to the ringbuffer `buf`.
*
* `in_samples` and `out_samples` define the rate conversion to perform on the
* samples in `data`. For negative rates, `out_samples` must be negative and
* `in_samples` positive.
*
* When `out_samples` is positive, the first sample will be written at position `sample`
* in the ringbuffer. When `out_samples` is negative, the last sample will be written to
* `sample` in reverse order.
*
* `out_samples` does not need to be a multiple of the segment size of the ringbuffer
* although it is recommended for optimal performance.
*
* `accum` will hold a temporary accumulator used in rate conversion and should be
* set to 0 when this function is first called. In case the commit operation is
* interrupted, one can resume the processing by passing the previously returned
* `accum` value back to this function.
*
* MT safe.
* @param sample the sample position of the data
* @param data the data to commit
* @param out_samples the number of samples to write to the ringbuffer
* @param accum accumulator for rate conversion.
* @returns The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than @out_samples when @buf was interrupted with a flush or stop.
*/
commit(
sample: number,
data: Uint8Array | string,
out_samples: number,
accum: number,
): [number, number, number];
/**
* Convert `src_val` in `src_fmt` to the equivalent value in `dest_fmt`. The result
* will be put in `dest_val`.
* @param src_fmt the source format
* @param src_val the source value
* @param dest_fmt the destination format
* @returns TRUE if the conversion succeeded.
*/
convert(src_fmt: Gst.Format | null, src_val: number, dest_fmt: Gst.Format | null): [boolean, number];
/**
* Get the number of samples queued in the audio device. This is
* usually less than the segment size but can be bigger when the
* implementation uses another internal buffer between the audio
* device.
*
* For playback ringbuffers this is the amount of samples transferred from the
* ringbuffer to the device but still not played.
*
* For capture ringbuffers this is the amount of samples in the device that are
* not yet transferred to the ringbuffer.
* @returns The number of samples queued in the audio device. MT safe.
*/
delay(): number;
/**
* Checks the status of the device associated with the ring buffer.
* @returns TRUE if the device was open, FALSE if it was closed. MT safe.
*/
device_is_open(): boolean;
/**
* Gets the current segment base number of the ringbuffer.
*
* MT safe.
* @returns Current segment base number of the ringbuffer.
*/
get_segbase(): number;
/**
* Gets the current segment number of the ringbuffer.
*
* MT safe.
* @returns Current segment number of the ringbuffer.
*/
get_segdone(): number;
/**
* Check if the ringbuffer is acquired and ready to use.
* @returns TRUE if the ringbuffer is acquired, FALSE on error. MT safe.
*/
is_acquired(): boolean;
/**
* Check if `buf` is activated.
*
* MT safe.
* @returns TRUE if the device is active.
*/
is_active(): boolean;
/**
* Check if `buf` is flushing.
*
* MT safe.
* @returns TRUE if the device is flushing.
*/
is_flushing(): boolean;
/**
* Tell the ringbuffer that it is allowed to start playback when
* the ringbuffer is filled with samples.
*
* MT safe.
* @param allowed the new value
*/
may_start(allowed: boolean): void;
/**
* Open the audio device associated with the ring buffer. Does not perform any
* setup on the device. You must open the device before acquiring the ring
* buffer.
* @returns TRUE if the device could be opened, FALSE on error. MT safe.
*/
open_device(): boolean;
/**
* Pause processing samples from the ringbuffer.
* @returns TRUE if the device could be paused, FALSE on error. MT safe.
*/
pause(): boolean;
/**
* Returns a pointer to memory where the data from segment `segment`
* can be found. This function is mostly used by subclasses.
* @returns FALSE if the buffer is not started. MT safe.
*/
prepare_read(): [boolean, number, Uint8Array];
/**
* Read `len` samples from the ringbuffer into the memory pointed
* to by `data`.
* The first sample should be read from position `sample` in
* the ringbuffer.
*
* `len` should not be a multiple of the segment size of the ringbuffer
* although it is recommended.
*
* `timestamp` will return the timestamp associated with the data returned.
* @param sample the sample position of the data
* @param data where the data should be read
* @returns The number of samples read from the ringbuffer or -1 on error. MT safe.
*/
read(sample: number, data: Uint8Array | string): [number, Gst.ClockTime];
/**
* Free the resources of the ringbuffer.
* @returns TRUE if the device could be released, FALSE on error. MT safe.
*/
release(): boolean;
/**
* Get the number of samples that were processed by the ringbuffer
* since it was last started. This does not include the number of samples not
* yet processed (see gst_audio_ring_buffer_delay()).
* @returns The number of samples processed by the ringbuffer. MT safe.
*/
samples_done(): number;
/**
* Sets the given callback function on the buffer. This function
* will be called every time a segment has been written to a device.
*
* MT safe.
* @param cb the callback to set
*/
set_callback(cb?: AudioRingBufferCallback | null): void;
/**
* Tell the ringbuffer about the device's channel positions. This must
* be called in when the ringbuffer is acquired.
* @param position the device channel positions
*/
set_channel_positions(position: AudioChannelPosition[] | null): void;
/**
* Mark the ringbuffer as errored after it has started.
*
* MT safe.
*/
set_errored(): void;
/**
* Set the ringbuffer to flushing mode or normal mode.
*
* MT safe.
* @param flushing the new mode
*/
set_flushing(flushing: boolean): void;
/**
* Make sure that the next sample written to the device is
* accounted for as being the `sample` sample written to the
* device. This value will be used in reporting the current
* sample position of the ringbuffer.
*
* This function will also clear the buffer with silence.
*
* MT safe.
* @param sample the sample number to set
*/
set_sample(sample: number): void;
/**
* Sets the current segment number of the ringbuffer.
*
* MT safe.
* @param segdone the segment number to set
*/
set_segdone(segdone: number): void;
set_timestamp(readseg: number, timestamp: Gst.ClockTime): void;
/**
* Start processing samples from the ringbuffer.
* @returns TRUE if the device could be started, FALSE on error. MT safe.
*/
start(): boolean;
/**
* Stop processing samples from the ringbuffer.
* @returns TRUE if the device could be stopped, FALSE on error. MT safe.
*/
stop(): boolean;
}
namespace AudioSink {
// Constructor properties interface
interface ConstructorProps extends AudioBaseSink.ConstructorProps {}
}
/**
* This is the most simple base class for audio sinks that only requires
* subclasses to implement a set of simple functions:
*
* * `open()` :Open the device.
*
* * `prepare()` :Configure the device with the specified format.
*
* * `write()` :Write samples to the device.
*
* * `reset()` :Unblock writes and flush the device.
*
* * `delay()` :Get the number of samples written but not yet played
* by the device.
*
* * `unprepare()` :Undo operations done by prepare.
*
* * `close()` :Close the device.
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstAudioBaseSink using a default implementation of a
* #GstAudioRingBuffer that uses threads.
*/
class AudioSink extends AudioBaseSink {
static $gtype: GObject.GType<AudioSink>;
// Fields
element: AudioBaseSink;
// Constructors
constructor(properties?: Partial<AudioSink.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* Close the device.
*/
vfunc_close(): boolean;
/**
* Return how many frames are still in the device. Participates in
* computing the time for audio clocks and drives the synchronisation.
*/
vfunc_delay(): number;
/**
* Open the device. No configuration needs to be done at this point.
* This function is also used to check if the device is available.
*/
vfunc_open(): boolean;
/**
* Pause the device and unblock write as fast as possible.
* For retro compatibility, the audio sink will fallback
* to calling reset if this vmethod is not provided. Since: 1.18
*/
vfunc_pause(): void;
/**
* Prepare the device to operate with the specified parameters.
* @param spec
*/
vfunc_prepare(spec: AudioRingBufferSpec): boolean;
// Conflicted with GstBase.BaseSink.vfunc_prepare
vfunc_prepare(...args: never[]): any;
/**
* Returns as quickly as possible from a write and flush any pending
* samples from the device.
* This vmethod is deprecated. Please provide pause and stop instead.
*/
vfunc_reset(): void;
/**
* Resume the device. Since: 1.18
*/
vfunc_resume(): void;
/**
* Stop the device and unblock write as fast as possible.
* Pending samples are flushed from the device.
* For retro compatibility, the audio sink will fallback
* to calling reset if this vmethod is not provided. Since: 1.18
*/
vfunc_stop(): void;
// Conflicted with GstBase.BaseSink.vfunc_stop
vfunc_stop(...args: never[]): any;
/**
* Undo operations done in prepare.
*/
vfunc_unprepare(): boolean;
/**
* Write samples to the device.
* @param data the sample data
*/
vfunc_write(data: Uint8Array | string): number;
}
namespace AudioSrc {
// Constructor properties interface
interface ConstructorProps extends AudioBaseSrc.ConstructorProps {}
}
/**
* This is the most simple base class for audio sources that only requires
* subclasses to implement a set of simple functions:
*
* * `open()` :Open the device.
* * `prepare()` :Configure the device with the specified format.
* * `read()` :Read samples from the device.
* * `reset()` :Unblock reads and flush the device.
* * `delay()` :Get the number of samples in the device but not yet read.
* * `unprepare()` :Undo operations done by prepare.
* * `close()` :Close the device.
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstAudioBaseSrc using a default implementation of a
* #GstAudioRingBuffer that uses threads.
*/
class AudioSrc extends AudioBaseSrc {
static $gtype: GObject.GType<AudioSrc>;
// Fields
element: AudioBaseSrc;
// Constructors
constructor(properties?: Partial<AudioSrc.ConstructorProps>, ...args: any[]);
_init(...args: any[]): void;
// Virtual methods
/**
* close the device
*/
vfunc_close(): boolean;
/**
* the number of frames queued in the device
*/
vfunc_delay(): number;
/**
* open the device with the specified caps
*/
vfunc_open(): boolean;
/**
* configure device with format
* @param spec
*/
vfunc_prepare(spec: AudioRingBufferSpec): boolean;
/**
* Read samples from the device.
* @param data the sample data
*/
vfunc_read(data: Uint8Array | string): [number, Gst.ClockTime];
/**
* unblock a read to the device and reset.
*/
vfunc_reset(): void;
/**
* undo the configuration
*/
vfunc_unprepare(): boolean;
}
type AudioAggregatorClass = typeof AudioAggregator;
type AudioAggregatorConvertPadClass = typeof AudioAggregatorConvertPad;
abstract class AudioAggregatorConvertPadPrivate {
static $gtype: GObject.GType<AudioAggregatorConvertPadPrivate>;
// Constructors
_init(...args: any[]): void;
}
type AudioAggregatorPadClass = typeof AudioAggregatorPad;
abstract class AudioAggregatorPadPrivate {
static $gtype: GObject.GType<AudioAggregatorPadPrivate>;
// Constructors
_init(...args: any[]): void;
}
abstract class AudioAggregatorPrivate {
static $gtype: GObject.GType<AudioAggregatorPrivate>;
// Constructors
_init(...args: any[]): void;
}
type AudioBaseSinkClass = typeof AudioBaseSink;
abstract class AudioBaseSinkPrivate {
static $gtype: GObject.GType<AudioBaseSinkPrivate>;
// Constructors
_init(...args: any[]): void;
}
type AudioBaseSrcClass = typeof AudioBaseSrc;
abstract class AudioBaseSrcPrivate {
static $gtype: GObject.GType<AudioBaseSrcPrivate>;
// Constructors
_init(...args: any[]): void;
}
/**
* A structure containing the result of an audio buffer map operation,
* which is executed with gst_audio_buffer_map(). For non-interleaved (planar)
* buffers, the beginning of each channel in the buffer has its own pointer in
* the `planes` array. For interleaved buffers, the `planes` array only contains
* one item, which is the pointer to the beginning of the buffer, and `n_planes`
* equals 1.
*
* The different channels in `planes` are always in the GStreamer channel order.
*/
class AudioBuffer {
static $gtype: GObject.GType<AudioBuffer>;
// Fields
n_samples: number;
n_planes: number;
planes: any;
// Constructors
_init(...args: any[]): void;
// Static methods
/**
* Clip the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* `buffer` anymore.
* @param buffer The buffer to clip.
* @param segment Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.
* @param rate sample rate.
* @param bpf size of one audio frame in bytes. This is the size of one sample * number of channels.
*/
static clip(buffer: Gst.Buffer, segment: Gst.Segment, rate: number, bpf: number): Gst.Buffer | null;
/**
* Maps an audio `gstbuffer` so that it can be read or written and stores the
* result of the map operation in `buffer`.
*
* This is especially useful when the `gstbuffer` is in non-interleaved (planar)
* layout, in which case this function will use the information in the
* `gstbuffer'`s attached #GstAudioMeta in order to map each channel in a
* separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached
* on the `gstbuffer,` then it must be in interleaved layout.
*
* If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked
* against `info`. Normally, they should be equal, but in case they are not,
* a g_critical will be printed and the #GstAudioInfo from the meta will be
* used.
*
* In non-interleaved buffers, it is possible to have each channel on a separate
* #GstMemory. In this case, each memory will be mapped separately to avoid
* copying their contents in a larger memory area. Do note though that it is
* not supported to have a single channel spanning over two or more different
* #GstMemory objects. Although the map operation will likely succeed in this
* case, it will be highly sub-optimal and it is recommended to merge all the
* memories in the buffer before calling this function.
*
* Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid
* as long as it's mapped.
* @param info the audio properties of the buffer
* @param gstbuffer the #GstBuffer to be mapped
* @param flags the access mode for the memory
*/
static map(info: AudioInfo, gstbuffer: Gst.Buffer, flags: Gst.MapFlags): [boolean, AudioBuffer];
/**
* Reorders `buffer` from the channel positions `from` to the channel
* positions `to`. `from` and `to` must contain the same number of
* positions and the same positions, only in a different order.
* `buffer` must be writable.
* @param buffer The buffer to reorder.
* @param format The %GstAudioFormat of the buffer.
* @param from The channel positions in the buffer.
* @param to The channel positions to convert to.
*/
static reorder_channels(
buffer: Gst.Buffer,
format: AudioFormat,
from: AudioChannelPosition[],
to: AudioChannelPosition[],
): boolean;
/**
* Truncate the buffer to finally have `samples` number of samples, removing
* the necessary amount of samples from the end and `trim` number of samples
* from the beginning.
*
* This function does not know the audio rate, therefore the caller is
* responsible for re-setting the correct timestamp and duration to the
* buffer. However, timestamp will be preserved if trim == 0, and duration
* will also be preserved if there is no trimming to be done. Offset and
* offset end will be preserved / updated.
*
* After calling this function the caller does not own a reference to
* `buffer` anymore.
* @param buffer The buffer to truncate.
* @param bpf size of one audio frame in bytes. This is the size of one sample * number of channels.
* @param trim the number of samples to remove from the beginning of the buffer
* @param samples the final number of samples that should exist in this buffer or -1 to use all the remaining samples if you are only removing samples from the beginning.
*/
static truncate(buffer: Gst.Buffer, bpf: number, trim: number, samples: number): Gst.Buffer;
// Methods
/**
* Unmaps an audio buffer that was previously mapped with
* gst_audio_buffer_map().
*/
unmap(): void;
}
type AudioCdSrcClass = typeof AudioCdSrc;
abstract class AudioCdSrcPrivate {
static $gtype: GObject.GType<AudioCdSrcPrivate>;
// Constructors
_init(...args: any[]): void;
}
/**
* CD track abstraction to communicate TOC entries to the base class.
*
* This structure is only for use by sub-classed in connection with
* gst_audio_cd_src_add_track().
*
* Applications will be informed of the available tracks via a TOC message
* on the pipeline's #GstBus instead.
*/
class AudioCdSrcTrack {
static $gtype: GObject.GType<AudioCdSrcTrack>;
// Fields
is_audio: boolean;
num: number;
start: number;
end: number;
// Constructors
_init(...args: any[]): void;
}
abstract class AudioChannelMixer {
static $gtype: GObject.GType<AudioChannelMixer>;
// Constructors
_init(...args: any[]): void;
// Methods
/**
* Free memory allocated by `mix`.
*/
free(): void;
/**
* Check if `mix` is in passthrough.
*
* Only N x N mix identity matrices are considered passthrough,
* this is determined by comparing the contents of the matrix
* with 0.0 and 1.0.
*
* As this is floating point comparisons, if the values have been
* generated, they should be rounded up or down by explicit
* assignment of 0.0 or 1.0 to values within a user-defined
* epsilon, this code doesn't make assumptions as to what may
* constitute an appropriate epsilon.
* @returns %TRUE is @mix is passthrough.
*/
is_passthrough(): boolean;
/**
* In case the samples are interleaved, `in` and `out` must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, `in` and `out` must point to an
* array with pointers to memory blocks, one for each channel.
*
* Perform channel mixing on `in_data` and write the result to `out_data`.
* `in_data` and `out_data` need to be in `format` and `layout`.
* @param _in input samples
* @param out output samples
* @param samples number of samples
*/
samples(_in: any | null, out: any | null, samples: number): void;
}
/**
* Extra buffer metadata describing how much audio has to be clipped from
* the start or end of a buffer. This is used for compressed formats, where
* the first frame usually has some additional samples due to encoder and
* decoder delays, and the last frame usually has some additional samples to
* be able to fill the complete last frame.
*
* This is used to ensure that decoded data in the end has the same amount of
* samples, and multiply decoded streams can be gaplessly concatenated.
*
* Note: If clipping of the start is done by adjusting the segment, this meta
* has to be dropped from buffers as otherwise clipping could happen twice.
*/
class AudioClippingMeta {
static $gtype: GObject.GType<AudioClippingMeta>;
// Fields
format: Gst.Format;
start: number;
end: number;
// Constructors
_init(...args: any[]): void;
// Static methods
static get_info(): Gst.MetaInfo;
}
type AudioClockClass = typeof AudioClock;
/**
* This object is used to convert audio samples from one format to another.
* The object can perform conversion of:
*
* * audio format with optional dithering and noise shaping
*
* * audio samplerate
*
* * audio channels and channel layout
*/
class AudioConverter {
static $gtype: GObject.GType<AudioConverter>;
// Constructors
constructor(
flags: AudioConverterFlags,
in_info: AudioInfo,
out_info: AudioInfo,
config?: Gst.Structure | null,
);
_init(...args: any[]): void;
static ['new'](
flags: AudioConverterFlags,
in_info: AudioInfo,
out_info: AudioInfo,
config?: Gst.Structure | null,
): AudioConverter;
// Methods
/**
* Convenience wrapper around gst_audio_converter_samples(), which will
* perform allocation of the output buffer based on the result from
* gst_audio_converter_get_out_frames().
* @param flags extra #GstAudioConverterFlags
* @param _in input data
* @returns %TRUE is the conversion could be performed.
*/
convert(flags: AudioConverterFlags | null, _in: Uint8Array | string): [boolean, Uint8Array];
/**
* Free a previously allocated `convert` instance.
*/
free(): void;
/**
* Get the current configuration of `convert`.
* @returns a #GstStructure that remains valid for as long as @convert is valid or until gst_audio_converter_update_config() is called.
*/
get_config(): [Gst.Structure, number, number];
/**
* Calculate how many input frames are currently needed by `convert` to produce
* `out_frames` of output frames.
* @param out_frames number of output frames
* @returns the number of input frames
*/
get_in_frames(out_frames: number): number;
/**
* Get the maximum number of input frames that the converter would
* need before producing output.
* @returns the latency of @convert as expressed in the number of frames.
*/
get_max_latency(): number;
/**
* Calculate how many output frames can be produced when `in_frames` input
* frames are given to `convert`.
* @param in_frames number of input frames
* @returns the number of output frames
*/
get_out_frames(in_frames: number): number;
/**
* Returns whether the audio converter will operate in passthrough mode.
* The return value would be typically input to gst_base_transform_set_passthrough()
* @returns %TRUE when no conversion will actually occur.
*/
is_passthrough(): boolean;
/**
* Reset `convert` to the state it was when it was first created, clearing
* any history it might currently have.
*/
reset(): void;
/**
* Perform the conversion with `in_frames` in `in` to `out_frames` in `out`
* using `convert`.
*
* In case the samples are interleaved, `in` and `out` must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, `in` and `out` must point to an
* array with pointers to memory blocks, one for each channel.
*
* `in` may be %NULL, in which case `in_frames` of silence samples are processed
* by the converter.
*
* This function always produces `out_frames` of output and consumes `in_frames` of
* input. Use gst_audio_converter_get_out_frames() and
* gst_audio_converter_get_in_frames() to make sure `in_frames` and `out_frames`
* are matching and `in` and `out` point to enough memory.
* @param flags extra #GstAudioConverterFlags
* @param _in input frames
* @param in_frames number of input frames
* @param out output frames
* @param out_frames number of output frames
* @returns %TRUE is the conversion could be performed.
*/
samples(
flags: AudioConverterFlags | null,
_in: any | null,
in_frames: number,
out: any | null,
out_frames: number,
): boolean;
/**
* Returns whether the audio converter can perform the conversion in-place.
* The return value would be typically input to gst_base_transform_set_in_place()
* @returns %TRUE when the conversion can be done in place.
*/
supports_inplace(): boolean;
/**
* Set `in_rate,` `out_rate` and `config` as extra configuration for `convert`.
*
* `in_rate` and `out_rate` specify the new sample rates of input and output
* formats. A value of 0 leaves the sample rate unchanged.
*
* `config` can be %NULL, in which case, the current configuration is not
* changed.
*
* If the parameters in `config` can not be set exactly, this function returns
* %FALSE and will try to update as much state as possible. The new state can
* then be retrieved and refined with gst_audio_converter_get_config().
*
* Look at the `GST_AUDIO_CONVERTER_OPT_*` fields to check valid configuration
* option and values.
* @param in_rate input rate
* @param out_rate output rate
* @param config a #GstStructure or %NULL
* @returns %TRUE when the new parameters could be set
*/
update_config(in_rate: number, out_rate: number, config?: Gst.Structure | null): boolean;
}
type AudioDecoderClass = typeof AudioDecoder;
abstract class AudioDecoderPrivate {
static $gtype: GObject.GType<AudioDecoderPrivate>;
// Constructors
_init(...args: any[]): void;
}
/**
* Extra buffer metadata describing audio downmixing matrix. This metadata is
* attached to audio buffers and contains a matrix to downmix the buffer number
* of channels to `channels`.
*
* `matrix` is an two-dimensional array of `to_channels` times `from_channels`
* coefficients, i.e. the i-th output channels is constructed by multiplicating
* the input channels with the coefficients in `matrix[`i] and taking the sum
* of the results.
*/
class AudioDownmixMeta {
static $gtype: GObject.GType<AudioDownmixMeta>;
// Fields
from_position: AudioChannelPosition;
to_position: AudioChannelPosition;
from_channels: number;
to_channels: number;
matrix: number;
// Constructors
_init(...args: any[]): void;
// Static methods
static get_info(): Gst.MetaInfo;
}
type AudioEncoderClass = typeof AudioEncoder;
abstract class AudioEncoderPrivate {
static $gtype: GObject.GType<AudioEncoderPrivate>;
// Constructors
_init(...args: any[]): void;
}
type AudioFilterClass = typeof AudioFilter;
/**
* Information for an audio format.
*/
class AudioFormatInfo {
static $gtype: GObject.GType<AudioFormatInfo>;
// Fields
format: AudioFormat;
name: string;
description: string;
flags: AudioFormatFlags;
endianness: number;
width: number;
depth: number;
silence: Uint8Array;
unpack_format: AudioFormat;
unpack_func: AudioFormatUnpack;
pack_func: AudioFormatPack;
// Constructors
_init(...args: any[]): void;
// Methods
/**
* Fill `length` bytes in `dest` with silence samples for `info`.
* @param dest a destination to fill
*/
fill_silence(dest: Uint8Array | string): void;
}
/**
* Information describing audio properties. This information can be filled
* in from GstCaps with gst_audio_info_from_caps().
*
* Use the provided macros to access the info in this structure.
*/
class AudioInfo {
static $gtype: GObject.GType<AudioInfo>;
// Fields
flags: AudioFlags;
layout: AudioLayout;
rate: number;
channels: number;
bpf: number;
position: AudioChannelPosition[];
// Constructors
constructor(
properties?: Partial<{
flags: AudioFlags;
layout: AudioLayout;
rate: number;
channels: number;
bpf: number;
position: AudioChannelPosition[];
}>,
);
_init(...args: any[]): void;
static ['new'](): AudioInfo;
static new_from_caps(caps: Gst.Caps): AudioInfo;
// Static methods
/**
* Parse `caps` and update `info`.
* @param caps a #GstCaps
*/
static from_caps(caps: Gst.Caps): [boolean, AudioInfo];
/**
* Initialize `info` with default values.
*/
static init(): AudioInfo;
// Methods
/**
* Converts among various #GstFormat types. This function handles
* GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
* raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
* function can be used to handle pad queries of the type GST_QUERY_CONVERT.
* @param src_fmt #GstFormat of the @src_val
* @param src_val value to convert
* @param dest_fmt #GstFormat of the @dest_val
* @returns TRUE if the conversion was successful.
*/
convert(src_fmt: Gst.Format | null, src_val: number, dest_fmt: Gst.Format | null): [boolean, number];
/**
* Copy a GstAudioInfo structure.
* @returns a new #GstAudioInfo. free with gst_audio_info_free.
*/
copy(): AudioInfo;
/**
* Free a GstAudioInfo structure previously allocated with gst_audio_info_new()
* or gst_audio_info_copy().
*/
free(): void;
/**
* Compares two #GstAudioInfo and returns whether they are equal or not
* @param other a #GstAudioInfo
* @returns %TRUE if @info and @other are equal, else %FALSE.
*/
is_equal(other: AudioInfo): boolean;
/**
* Set the default info for the audio info of `format` and `rate` and `channels`.
*
* Note: This initializes `info` first, no values are preserved.
* @param format the format
* @param rate the samplerate
* @param channels the number of channels
* @param position the channel positions
*/
set_format(
format: AudioFormat | null,
rate: number,
channels: number,
position?: AudioChannelPosition[] | null,
): void;
/**
* Convert the values of `info` into a #GstCaps.
* @returns the new #GstCaps containing the info of @info.
*/
to_caps(): Gst.Caps;
}
/**
* Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464
*/
class AudioLevelMeta {
static $gtype: GObject.GType<AudioLevelMeta>;
// Fields
level: number;
voice_activity: boolean;
// Constructors
_init(...args: any[]): void;
// Static methods
/**
* Return the #GstMetaInfo associated with #GstAudioLevelMeta.
*/
static get_info(): Gst.MetaInfo;
}
/**
* #GstAudioDownmixMeta defines an audio downmix matrix to be send along with
* audio buffers. These functions in this module help to create and attach the
* meta as well as extracting it.
*/
class AudioMeta {
static $gtype: GObject.GType<AudioMeta>;
// Fields
samples: number;
offsets: number;
// Constructors
_init(...args: any[]): void;
// Static methods
static get_info(): Gst.MetaInfo;
}
abstract class AudioQuantize {
static $gtype: GObject.GType<AudioQuantize>;
// Constructors
_init(...args: any[]): void;
// Methods
/**
* Free a #GstAudioQuantize.
*/
free(): void;
/**
* Reset `quant` to the state is was when created, clearing any
* history it might have.
*/
reset(): void;
/**
* Perform quantization on `samples` in `in` and write the result to `out`.
*
* In case the samples are interleaved, `in` and `out` must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, `in` and `out` must point to an
* array with pointers to memory blocks, one for each channel.
*
* `in` and `out` may point to the same memory location, in which case samples will be
* modified in-place.
* @param _in input samples
* @param out output samples
* @param samples number of samples
*/
samples(_in: any | null, out: any | null, samples: number): void;
}
/**
* #GstAudioResampler is a structure which holds the information
* required to perform various kinds of resampling filtering.
*/
abstract class AudioResampler {
static $gtype: GObject.GType<AudioResampler>;
// Constructors
_init(...args: any[]): void;
// Static methods
/**
* Make a new resampler.
* @param method a #GstAudioResamplerMethod
* @param flags #GstAudioResamplerFlags
* @param format the #GstAudioFormat
* @param channels the number of channels
* @param in_rate input rate
* @param out_rate output rate
* @param options extra options
*/
static ['new'](
method: AudioResamplerMethod,
flags: AudioResamplerFlags,
format: AudioFormat,
channels: number,
in_rate: number,
out_rate: number,
options: Gst.Structure,
): AudioResampler;
/**
* Set the parameters for resampling from `in_rate` to `out_rate` using `method`
* for `quality` in `options`.
* @param method a #GstAudioResamplerMethod
* @param quality the quality
* @param in_rate the input rate
* @param out_rate the output rate
* @param options a #GstStructure
*/
static options_set_quality(
method: AudioResamplerMethod,
quality: number,
in_rate: number,
out_rate: number,
options: Gst.Structure,
): void;
// Methods
/**
* Free a previously allocated #GstAudioResampler `resampler`.
*/
free(): void;
/**
* Get the number of input frames that would currently be needed
* to produce `out_frames` from `resampler`.
* @param out_frames number of input frames
* @returns The number of input frames needed for producing @out_frames of data from @resampler.
*/
get_in_frames(out_frames: number): number;
/**
* Get the maximum number of input samples that the resampler would
* need before producing output.
* @returns the latency of @resampler as expressed in the number of frames.
*/
get_max_latency(): number;
/**
* Get the number of output frames that would be currently available when
* `in_frames` are given to `resampler`.
* @param in_frames number of input frames
* @returns The number of frames that would be available after giving @in_frames as input to @resampler.
*/
get_out_frames(in_frames: number): number;
/**
* Perform resampling on `in_frames` frames in `in` and write `out_frames` to `out`.
*
* In case the samples are interleaved, `in` and `out` must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, `in` and `out` must point to an
* array with pointers to memory blocks, one for each channel.
*
* `in` may be %NULL, in which case `in_frames` of silence samples are pushed
* into the resampler.
*
* This function always produces `out_frames` of output and consumes `in_frames` of
* input. Use gst_audio_resampler_get_out_frames() and
* gst_audio_resampler_get_in_frames() to make sure `in_frames` and `out_frames`
* are matching and `in` and `out` point to enough memory.
* @param _in input samples
* @param in_frames number of input frames
* @param out output samples
* @param out_frames number of output frames
*/
resample(_in: any | null, in_frames: number, out: any | null, out_frames: number): void;
/**
* Reset `resampler` to the state it was when it was first created, discarding
* all sample history.
*/
reset(): void;
/**
* Update the resampler parameters for `resampler`. This function should
* not be called concurrently with any other function on `resampler`.
*
* When `in_rate` or `out_rate` is 0, its value is unchanged.
*
* When `options` is %NULL, the previously configured options are reused.
* @param in_rate new input rate
* @param out_rate new output rate
* @param options new options or %NULL
* @returns %TRUE if the new parameters could be set
*/
update(in_rate: number, out_rate: number, options: Gst.Structure): boolean;
}
type AudioRingBufferClass = typeof AudioRingBuffer;
abstract class AudioRingBufferPrivate {
static $gtype: GObject.GType<AudioRingBufferPrivate>;
// Constructors
_init(...args: any[]): void;
}
/**
* The structure containing the format specification of the ringbuffer.
*
* When `type` is GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DSD, the `dsd_format`
* is valid (otherwise it is unused). Also, when DSD is the sample type,
* only the rate, channels, position, and bpf fields in `info` are populated.
*/
class AudioRingBufferSpec {
static $gtype: GObject.GType<AudioRingBufferSpec>;
// Fields
type: AudioRingBufferFormatType;
latency_time: number;
buffer_time: number;
segsize: number;
segtotal: number;
seglatency: number;
// Constructors
_init(...args: any[]): void;
}
type AudioSinkClass = typeof AudioSink;
class AudioSinkClassExtension {
static $gtype: GObject.GType<AudioSinkClassExtension>;
// Constructors
_init(...args: any[]): void;
}
type AudioSrcClass = typeof AudioSrc;
/**
* #GstAudioStreamAlign provides a helper object that helps tracking audio
* stream alignment and discontinuities, and detects discontinuities if
* possible.
*
* See gst_audio_stream_align_new() for a description of its parameters and
* gst_audio_stream_align_process() for the details of the processing.
*/
class AudioStreamAlign {
static $gtype: GObject.GType<AudioStreamAlign>;
// Constructors
constructor(rate: number, alignment_threshold: Gst.ClockTime, discont_wait: Gst.ClockTime);
_init(...args: any[]): void;
static ['new'](
rate: number,
alignment_threshold: Gst.ClockTime,
discont_wait: Gst.ClockTime,
): AudioStreamAlign;
// Methods
/**
* Copy a GstAudioStreamAlign structure.
* @returns a new #GstAudioStreamAlign. free with gst_audio_stream_align_free.
*/
copy(): AudioStreamAlign;
/**
* Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new()
* or gst_audio_stream_align_copy().
*/
free(): void;
/**
* Gets the currently configured alignment threshold.
* @returns The currently configured alignment threshold
*/
get_alignment_threshold(): Gst.ClockTime;
/**
* Gets the currently configured discont wait.
* @returns The currently configured discont wait
*/
get_discont_wait(): Gst.ClockTime;
/**
* Gets the currently configured sample rate.
* @returns The currently configured sample rate
*/
get_rate(): number;
/**
* Returns the number of samples that were processed since the last
* discontinuity was detected.
* @returns The number of samples processed since the last discontinuity.
*/
get_samples_since_discont(): number;
/**
* Timestamp that was passed when a discontinuity was detected, i.e. the first
* timestamp after the discontinuity.
* @returns The last timestamp at when a discontinuity was detected
*/
get_timestamp_at_discont(): Gst.ClockTime;
/**
* Marks the next buffer as discontinuous and resets timestamp tracking.
*/
mark_discont(): void;
/**
* Processes data with `timestamp` and `n_samples,` and returns the output
* timestamp, duration and sample position together with a boolean to signal
* whether a discontinuity was detected or not. All non-discontinuous data
* will have perfect timestamps and durations.
*
* A discontinuity is detected once the difference between the actual
* timestamp and the timestamp calculated from the sample count since the last
* discontinuity differs by more than the alignment threshold for a duration
* longer than discont wait.
*
* Note: In reverse playback, every buffer is considered discontinuous in the
* context of buffer flags because the last sample of the previous buffer is
* discontinuous with the first sample of the current one. However for this
* function they are only considered discontinuous in reverse playback if the
* first sample of the previous buffer is discontinuous with the last sample
* of the current one.
* @param discont if this data is considered to be discontinuous
* @param timestamp a #GstClockTime of the start of the data
* @param n_samples number of samples to process
* @returns %TRUE if a discontinuity was detected, %FALSE otherwise.
*/
process(
discont: boolean,
timestamp: Gst.ClockTime,
n_samples: number,
): [boolean, Gst.ClockTime, Gst.ClockTime, number];
/**
* Sets `alignment_treshold` as new alignment threshold for the following processing.
* @param alignment_threshold a new alignment threshold
*/
set_alignment_threshold(alignment_threshold: Gst.ClockTime): void;
/**
* Sets `alignment_treshold` as new discont wait for the following processing.
* @param discont_wait a new discont wait
*/
set_discont_wait(discont_wait: Gst.ClockTime): void;
/**
* Sets `rate` as new sample rate for the following processing. If the sample
* rate differs this implicitly marks the next data as discontinuous.
* @param rate a new sample rate
*/
set_rate(rate: number): void;
}
/**
* Information describing DSD audio properties.
*
* In DSD, the "sample format" is the bit. Unlike PCM, there are no further
* "sample formats" in DSD. However, in software, DSD bits are grouped into
* bytes (since dealing with individual bits is impractical), and these bytes
* in turn are grouped into words. This becomes relevant when interleaving
* channels and transmitting DSD data through audio APIs. The different
* types of grouping DSD bytes are referred to as the "DSD grouping forma"
* or just "DSD format". #GstDsdFormat has a list of valid ways of grouping
* DSD bytes into words.
*
* DSD rates are equivalent to PCM sample rates, except that they specify
* how many DSD bytes are consumed per second. This refers to the bytes per
* second _per channel_; the rate does not change when the number of channel
* changes. (Strictly speaking, it would be more correct to measure the
* *bits* per second, since the bit is the DSD "sample format", but it is
* more practical to use bytes.) In DSD, bit rates are always an integer
* multiple of the CD audio rate (44100) or the DAT rate (48000). DSD64-44x
* is 44100 * 64 = 2822400 bits per second, or 352800 bytes per second
* (the latter would be used in this info structure). DSD64-48x is
* 48000 * 64 = 3072000 bits per second, or 384000 bytes per second.
* #GST_DSD_MAKE_DSD_RATE_44x can be used for specifying DSD-44x rates,
* *and #GST_DSD_MAKE_DSD_RATE_48x can be used for specifying DSD-48x ones.
* Also, since DSD-48x is less well known, when the multiplier is given
* without the 44x/48x specifier, 44x is typically implied.
*
* It is important to know that in DSD, different format widths correspond
* to different playtimes. That is, a word with 32 DSD bits covers two times
* as much playtime as a word with 16 DSD bits. This is in contrast to PCM,
* where one word (= one PCM sample) always covers a time period of 1/samplerate,
* no matter how many bits a PCM sample is made of. For this reason, DSD
* and PCM widths and strides cannot be used the same way.
*
* Multiple channels are arranged in DSD data either interleaved or non-
* interleaved. This is similar to PCM. Interleaved layouts rotate between
* channels and words. First, word 0 of channel 0 is present. Then word
* 0 of channel 1 follows. Then word 0 of channel 2 etc. until all
* channels are through, then comes word 1 of channel 0 etc.
*
* Non-interleaved data is planar. First, all words of channel 0 are
* present, then all words of channel 1 etc. Unlike interleaved data,
* non-interleaved data can be sparse, that is, there can be space in
* between the planes. the `positions` array specifies the plane offsets.
*
* In uncommon cases, the DSD bits in the data bytes can be stored in reverse
* order. For example, normally, in DSDU8, the first byte contains DSD bits
* 0 to 7, and the most significant bit of that byte is DSD bit 0. If this
* order is reversed, then bit 7 is the first one instead. In that ase,
* `reversed_bytes` is set to TRUE.
*
* Use the provided macros to access the info in this structure.
*/
class DsdInfo {
static $gtype: GObject.GType<DsdInfo>;
// Fields
format: DsdFormat;
rate: number;
channels: number;
layout: AudioLayout;
reversed_bytes: boolean;
positions: AudioChannelPosition[];
flags: AudioFlags;
// Constructors
constructor(
properties?: Partial<{
format: DsdFormat;
rate: number;
channels: number;
layout: AudioLayout;
reversed_bytes: boolean;
positions: AudioChannelPosition[];
flags: AudioFlags;
}>,
);
_init(...args: any[]): void;
static ['new'](): DsdInfo;
static new_from_caps(caps: Gst.Caps): DsdInfo;
// Static methods
/**
* Parse `caps` and update `info`.
* @param caps a #GstCaps
*/
static from_caps(caps: Gst.Caps): [boolean, DsdInfo];
/**
* Initialize `info` with default values.
*/
static init(): DsdInfo;
// Methods
/**
* Copy a GstDsdInfo structure.
* @returns a new #GstDsdInfo. free with gst_dsd_info_free.
*/
copy(): DsdInfo;
/**
* Free a GstDsdInfo structure previously allocated with gst_dsd_info_new()
* or gst_dsd_info_copy().
*/
free(): void;
/**
* Compares two #GstDsdInfo and returns whether they are equal or not
* @param other a #GstDsdInfo
* @returns %TRUE if @info and @other are equal, else %FALSE.
*/
is_equal(other: DsdInfo): boolean;
/**
* Set the default info for the DSD info of `format` and `rate` and `channels`.
*
* Note: This initializes `info` first, no values are preserved.
* @param format the format
* @param rate the DSD rate
* @param channels the number of channels
* @param positions the channel positions
*/
set_format(
format: DsdFormat | null,
rate: number,
channels: number,
positions?: AudioChannelPosition[] | null,
): void;
/**
* Convert the values of `info` into a #GstCaps.
* @returns the new #GstCaps containing the info of @info.
*/
to_caps(): Gst.Caps;
}
/**
* Buffer metadata describing planar DSD contents in the buffer. This is not needed
* for interleaved DSD data, and is required for non-interleaved (= planar) data.
*
* The different channels in `offsets` are always in the GStreamer channel order.
* Zero-copy channel reordering can be implemented by swapping the values in
* `offsets`.
*
* It is not allowed for channels to overlap in memory,
* i.e. for each i in [0, channels), the range
* [`offsets[`i], `offsets[`i] + `num_bytes_per_channel)` must not overlap
* with any other such range.
*
* It is, however, allowed to have parts of the buffer memory unused, by using
* `offsets` and `num_bytes_per_channel` in such a way that leave gaps on it.
* This is used to implement zero-copy clipping in non-interleaved buffers.
*
* Obviously, due to the above, it is not safe to infer the
* number of valid bytes from the size of the buffer. You should always
* use the `num_bytes_per_channel` variable of this metadata.
*/
class DsdPlaneOffsetMeta {
static $gtype: GObject.GType<DsdPlaneOffsetMeta>;
// Fields
num_channels: number;
num_bytes_per_channel: number;
offsets: number;
// Constructors
_init(...args: any[]): void;
// Static methods
static get_info(): Gst.MetaInfo;
}
type StreamVolumeInterface = typeof StreamVolume;
namespace StreamVolume {
// Constructor properties interface
interface ConstructorProps extends GObject.Object.ConstructorProps {
mute: boolean;
volume: number;
}
}
export interface StreamVolumeNamespace {
$gtype: GObject.GType<StreamVolume>;
prototype: StreamVolume;
convert_volume(from: StreamVolumeFormat, to: StreamVolumeFormat, val: number): number;
}
interface StreamVolume extends GObject.Object {
// Properties
get mute(): boolean;
set mute(val: boolean);
get volume(): number;
set volume(val: number);
// Methods
get_mute(): boolean;
get_volume(format: StreamVolumeFormat | null): number;
set_mute(mute: boolean): void;
set_volume(format: StreamVolumeFormat | null, val: number): void;
}
export const StreamVolume: StreamVolumeNamespace & {
new (): StreamVolume; // This allows `obj instanceof StreamVolume`
};
/**
* Name of the imported GIR library
* `see` https://gitlab.gnome.org/GNOME/gjs/-/blob/master/gi/ns.cpp#L188
*/
const __name__: string;
/**
* Version of the imported GIR library
* `see` https://gitlab.gnome.org/GNOME/gjs/-/blob/master/gi/ns.cpp#L189
*/
const __version__: string;
}
export default GstAudio;
}
declare module 'gi://GstAudio' {
import GstAudio10 from 'gi://GstAudio?version=1.0';
export default GstAudio10;
}
// END
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