GstAudioResampler

GstAudioResampler is a structure which holds the information required to perform various kinds of resampling filtering.

GstAudioResampler


GstAudio.AudioResampler


GstAudio.AudioResampler


Methods

gst_audio_resampler_free

gst_audio_resampler_free (GstAudioResampler * resampler)

Free a previously allocated GstAudioResampler resampler.

Parameters:

resampler

a GstAudioResampler

Since : 1.6


GstAudio.AudioResampler.free

def GstAudio.AudioResampler.free (self):
    #python wrapper for 'gst_audio_resampler_free'

Free a previously allocated GstAudio.AudioResampler resampler.

Parameters:

Since : 1.6


GstAudio.AudioResampler.prototype.free

function GstAudio.AudioResampler.prototype.free(): {
    // javascript wrapper for 'gst_audio_resampler_free'
}

Free a previously allocated GstAudio.AudioResampler resampler.

Parameters:

Since : 1.6


gst_audio_resampler_get_in_frames

gsize
gst_audio_resampler_get_in_frames (GstAudioResampler * resampler,
                                   gsize out_frames)

Get the number of input frames that would currently be needed to produce out_frames from resampler.

Parameters:

resampler

a GstAudioResampler

out_frames

number of input frames

Returns

The number of input frames needed for producing out_frames of data from resampler.


GstAudio.AudioResampler.get_in_frames

def GstAudio.AudioResampler.get_in_frames (self, out_frames):
    #python wrapper for 'gst_audio_resampler_get_in_frames'

Get the number of input frames that would currently be needed to produce out_frames from resampler.

Parameters:

out_frames ( int ) –

number of input frames

Returns ( int ) –

The number of input frames needed for producing out_frames of data from resampler.


GstAudio.AudioResampler.prototype.get_in_frames

function GstAudio.AudioResampler.prototype.get_in_frames(out_frames: Number): {
    // javascript wrapper for 'gst_audio_resampler_get_in_frames'
}

Get the number of input frames that would currently be needed to produce out_frames from resampler.

Parameters:

out_frames ( Number ) –

number of input frames

Returns ( Number ) –

The number of input frames needed for producing out_frames of data from resampler.


gst_audio_resampler_get_max_latency

gsize
gst_audio_resampler_get_max_latency (GstAudioResampler * resampler)

Get the maximum number of input samples that the resampler would need before producing output.

Parameters:

resampler

a GstAudioResampler

Returns

the latency of resampler as expressed in the number of frames.


GstAudio.AudioResampler.get_max_latency

def GstAudio.AudioResampler.get_max_latency (self):
    #python wrapper for 'gst_audio_resampler_get_max_latency'

Get the maximum number of input samples that the resampler would need before producing output.

Parameters:

Returns ( int ) –

the latency of resampler as expressed in the number of frames.


GstAudio.AudioResampler.prototype.get_max_latency

function GstAudio.AudioResampler.prototype.get_max_latency(): {
    // javascript wrapper for 'gst_audio_resampler_get_max_latency'
}

Get the maximum number of input samples that the resampler would need before producing output.

Parameters:

Returns ( Number ) –

the latency of resampler as expressed in the number of frames.


gst_audio_resampler_get_out_frames

gsize
gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
                                    gsize in_frames)

Get the number of output frames that would be currently available when in_frames are given to resampler.

Parameters:

resampler

a GstAudioResampler

in_frames

number of input frames

Returns

The number of frames that would be available after giving in_frames as input to resampler.


GstAudio.AudioResampler.get_out_frames

def GstAudio.AudioResampler.get_out_frames (self, in_frames):
    #python wrapper for 'gst_audio_resampler_get_out_frames'

Get the number of output frames that would be currently available when in_frames are given to resampler.

Parameters:

in_frames ( int ) –

number of input frames

Returns ( int ) –

The number of frames that would be available after giving in_frames as input to resampler.


GstAudio.AudioResampler.prototype.get_out_frames

function GstAudio.AudioResampler.prototype.get_out_frames(in_frames: Number): {
    // javascript wrapper for 'gst_audio_resampler_get_out_frames'
}

Get the number of output frames that would be currently available when in_frames are given to resampler.

Parameters:

in_frames ( Number ) –

number of input frames

Returns ( Number ) –

The number of frames that would be available after giving in_frames as input to resampler.


gst_audio_resampler_resample

gst_audio_resampler_resample (GstAudioResampler * resampler,
                              gpointer * in,
                              gsize in_frames,
                              gpointer * out,
                              gsize out_frames)

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.

Parameters:

resampler

a GstAudioResampler

in

input samples

in_frames

number of input frames

out

output samples

out_frames

number of output frames


GstAudio.AudioResampler.resample

def GstAudio.AudioResampler.resample (self, in, in_frames, out, out_frames):
    #python wrapper for 'gst_audio_resampler_resample'

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 None, 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 GstAudio.AudioResampler.get_out_frames and GstAudio.AudioResampler.get_in_frames to make sure in_frames and out_frames are matching and in and out point to enough memory.

Parameters:

in ( object ) –

input samples

in_frames ( int ) –

number of input frames

out ( object ) –

output samples

out_frames ( int ) –

number of output frames


GstAudio.AudioResampler.prototype.resample

function GstAudio.AudioResampler.prototype.resample(in: Object, in_frames: Number, out: Object, out_frames: Number): {
    // javascript wrapper for 'gst_audio_resampler_resample'
}

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 GstAudio.AudioResampler.prototype.get_out_frames and GstAudio.AudioResampler.prototype.get_in_frames to make sure in_frames and out_frames are matching and in and out point to enough memory.

Parameters:

in ( Object ) –

input samples

in_frames ( Number ) –

number of input frames

out ( Object ) –

output samples

out_frames ( Number ) –

number of output frames


gst_audio_resampler_reset

gst_audio_resampler_reset (GstAudioResampler * resampler)

Reset resampler to the state it was when it was first created, discarding all sample history.

Parameters:

resampler

a GstAudioResampler


GstAudio.AudioResampler.reset

def GstAudio.AudioResampler.reset (self):
    #python wrapper for 'gst_audio_resampler_reset'

Reset resampler to the state it was when it was first created, discarding all sample history.

Parameters:


GstAudio.AudioResampler.prototype.reset

function GstAudio.AudioResampler.prototype.reset(): {
    // javascript wrapper for 'gst_audio_resampler_reset'
}

Reset resampler to the state it was when it was first created, discarding all sample history.

Parameters:


gst_audio_resampler_update

gboolean
gst_audio_resampler_update (GstAudioResampler * resampler,
                            gint in_rate,
                            gint out_rate,
                            GstStructure * options)

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.

Parameters:

resampler

a GstAudioResampler

in_rate

new input rate

out_rate

new output rate

options

new options or NULL

Returns

TRUE if the new parameters could be set


GstAudio.AudioResampler.update

def GstAudio.AudioResampler.update (self, in_rate, out_rate, options):
    #python wrapper for 'gst_audio_resampler_update'

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 None, the previously configured options are reused.

Parameters:

in_rate ( int ) –

new input rate

out_rate ( int ) –

new output rate

options ( Gst.Structure ) –

new options or None

Returns ( bool ) –

True if the new parameters could be set


GstAudio.AudioResampler.prototype.update

function GstAudio.AudioResampler.prototype.update(in_rate: Number, out_rate: Number, options: Gst.Structure): {
    // javascript wrapper for 'gst_audio_resampler_update'
}

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.

Parameters:

in_rate ( Number ) –

new input rate

out_rate ( Number ) –

new output rate

options ( Gst.Structure ) –

new options or null

Returns ( Number ) –

true if the new parameters could be set


Functions

gst_audio_resampler_new

GstAudioResampler *
gst_audio_resampler_new (GstAudioResamplerMethod method,
                         GstAudioResamplerFlags flags,
                         GstAudioFormat format,
                         gint channels,
                         gint in_rate,
                         gint out_rate,
                         GstStructure * options)

Make a new resampler.

Parameters:

format

the GstAudioFormat

channels

the number of channels

in_rate

input rate

out_rate

output rate

options

extra options

Returns ( [transfer: full] ) –

The new GstAudioResampler, or NULL on failure.


GstAudio.audio_resampler_new

def GstAudio.audio_resampler_new (method, flags, format, channels, in_rate, out_rate, options):
    #python wrapper for 'gst_audio_resampler_new'

Make a new resampler.

Parameters:

channels ( int ) –

the number of channels

in_rate ( int ) –

input rate

out_rate ( int ) –

output rate

options ( Gst.Structure ) –

extra options

Returns ( GstAudio.AudioResampler ) –

The new GstAudio.AudioResampler, or None on failure.


GstAudio.prototype.audio_resampler_new

function GstAudio.prototype.audio_resampler_new(method: GstAudio.AudioResamplerMethod, flags: GstAudio.AudioResamplerFlags, format: GstAudio.AudioFormat, channels: Number, in_rate: Number, out_rate: Number, options: Gst.Structure): {
    // javascript wrapper for 'gst_audio_resampler_new'
}

Make a new resampler.

Parameters:

channels ( Number ) –

the number of channels

in_rate ( Number ) –

input rate

out_rate ( Number ) –

output rate

options ( Gst.Structure ) –

extra options

Returns ( GstAudio.AudioResampler ) –

The new GstAudio.AudioResampler, or null on failure.


gst_audio_resampler_options_set_quality

gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
                                         guint quality,
                                         gint in_rate,
                                         gint out_rate,
                                         GstStructure * options)

Set the parameters for resampling from in_rate to out_rate using method for quality in options.

Parameters:

quality

the quality

in_rate

the input rate

out_rate

the output rate

options

a GstStructure


GstAudio.audio_resampler_options_set_quality

def GstAudio.audio_resampler_options_set_quality (method, quality, in_rate, out_rate, options):
    #python wrapper for 'gst_audio_resampler_options_set_quality'

Set the parameters for resampling from in_rate to out_rate using method for quality in options.

Parameters:

quality ( int ) –

the quality

in_rate ( int ) –

the input rate

out_rate ( int ) –

the output rate

options ( Gst.Structure ) –

a Gst.Structure


GstAudio.prototype.audio_resampler_options_set_quality

function GstAudio.prototype.audio_resampler_options_set_quality(method: GstAudio.AudioResamplerMethod, quality: Number, in_rate: Number, out_rate: Number, options: Gst.Structure): {
    // javascript wrapper for 'gst_audio_resampler_options_set_quality'
}

Set the parameters for resampling from in_rate to out_rate using method for quality in options.

Parameters:

quality ( Number ) –

the quality

in_rate ( Number ) –

the input rate

out_rate ( Number ) –

the output rate

options ( Gst.Structure ) –

a Gst.Structure


Enumerations

GstAudioResamplerFilterInterpolation

The different filter interpolation methods.

Members

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE (0) –

no interpolation

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR (1) –

linear interpolation of the filter coefficients.

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC (2) –

cubic interpolation of the filter coefficients.


GstAudio.AudioResamplerFilterInterpolation

The different filter interpolation methods.

Members

GstAudio.AudioResamplerFilterInterpolation.NONE (0) –

no interpolation

GstAudio.AudioResamplerFilterInterpolation.LINEAR (1) –

linear interpolation of the filter coefficients.

GstAudio.AudioResamplerFilterInterpolation.CUBIC (2) –

cubic interpolation of the filter coefficients.


GstAudio.AudioResamplerFilterInterpolation

The different filter interpolation methods.

Members

GstAudio.AudioResamplerFilterInterpolation.NONE (0) –

no interpolation

GstAudio.AudioResamplerFilterInterpolation.LINEAR (1) –

linear interpolation of the filter coefficients.

GstAudio.AudioResamplerFilterInterpolation.CUBIC (2) –

cubic interpolation of the filter coefficients.


GstAudioResamplerFilterMode

Select for the filter tables should be set up.

Members

GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED (0) –

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.

GST_AUDIO_RESAMPLER_FILTER_MODE_FULL (1) –

Use full filter table. This uses more memory but less CPU.

GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO (2) –

Automatically choose between interpolated and full filter tables.


GstAudio.AudioResamplerFilterMode

Select for the filter tables should be set up.

Members

GstAudio.AudioResamplerFilterMode.INTERPOLATED (0) –

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 GstAudio.AudioResampler.update.

GstAudio.AudioResamplerFilterMode.FULL (1) –

Use full filter table. This uses more memory but less CPU.

GstAudio.AudioResamplerFilterMode.AUTO (2) –

Automatically choose between interpolated and full filter tables.


GstAudio.AudioResamplerFilterMode

Select for the filter tables should be set up.

Members

GstAudio.AudioResamplerFilterMode.INTERPOLATED (0) –

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 GstAudio.AudioResampler.prototype.update.

GstAudio.AudioResamplerFilterMode.FULL (1) –

Use full filter table. This uses more memory but less CPU.

GstAudio.AudioResamplerFilterMode.AUTO (2) –

Automatically choose between interpolated and full filter tables.


GstAudioResamplerFlags

Different resampler flags.

Members

GST_AUDIO_RESAMPLER_FLAG_NONE (0) –

no flags

GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN (1) –

input samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT (2) –

output samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE (4) –

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.


GstAudio.AudioResamplerFlags

Different resampler flags.

Members

GstAudio.AudioResamplerFlags.NONE (0) –

no flags

GstAudio.AudioResamplerFlags.NON_INTERLEAVED_IN (1) –

input samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GstAudio.AudioResamplerFlags.NON_INTERLEAVED_OUT (2) –

output samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GstAudio.AudioResamplerFlags.VARIABLE_RATE (4) –

optimize for dynamic updates of the sample rates with GstAudio.AudioResampler.update. This will select an interpolating filter when GstAudio.AudioResamplerFilterMode.AUTO is configured.


GstAudio.AudioResamplerFlags

Different resampler flags.

Members

GstAudio.AudioResamplerFlags.NONE (0) –

no flags

GstAudio.AudioResamplerFlags.NON_INTERLEAVED_IN (1) –

input samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GstAudio.AudioResamplerFlags.NON_INTERLEAVED_OUT (2) –

output samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

GstAudio.AudioResamplerFlags.VARIABLE_RATE (4) –

optimize for dynamic updates of the sample rates with GstAudio.AudioResampler.prototype.update. This will select an interpolating filter when GstAudio.AudioResamplerFilterMode.AUTO is configured.


GstAudioResamplerMethod

Different subsampling and upsampling methods

Members

GST_AUDIO_RESAMPLER_METHOD_NEAREST (0) –

Duplicates the samples when upsampling and drops when downsampling

GST_AUDIO_RESAMPLER_METHOD_LINEAR (1) –

Uses linear interpolation to reconstruct missing samples and averaging to downsample

GST_AUDIO_RESAMPLER_METHOD_CUBIC (2) –

Uses cubic interpolation

GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL (3) –

Uses Blackman-Nuttall windowed sinc interpolation

GST_AUDIO_RESAMPLER_METHOD_KAISER (4) –

Uses Kaiser windowed sinc interpolation


GstAudio.AudioResamplerMethod

Different subsampling and upsampling methods

Members

GstAudio.AudioResamplerMethod.NEAREST (0) –

Duplicates the samples when upsampling and drops when downsampling

GstAudio.AudioResamplerMethod.LINEAR (1) –

Uses linear interpolation to reconstruct missing samples and averaging to downsample

GstAudio.AudioResamplerMethod.CUBIC (2) –

Uses cubic interpolation

GstAudio.AudioResamplerMethod.BLACKMAN_NUTTALL (3) –

Uses Blackman-Nuttall windowed sinc interpolation

GstAudio.AudioResamplerMethod.KAISER (4) –

Uses Kaiser windowed sinc interpolation


GstAudio.AudioResamplerMethod

Different subsampling and upsampling methods

Members

GstAudio.AudioResamplerMethod.NEAREST (0) –

Duplicates the samples when upsampling and drops when downsampling

GstAudio.AudioResamplerMethod.LINEAR (1) –

Uses linear interpolation to reconstruct missing samples and averaging to downsample

GstAudio.AudioResamplerMethod.CUBIC (2) –

Uses cubic interpolation

GstAudio.AudioResamplerMethod.BLACKMAN_NUTTALL (3) –

Uses Blackman-Nuttall windowed sinc interpolation

GstAudio.AudioResamplerMethod.KAISER (4) –

Uses Kaiser windowed sinc interpolation


Constants

GST_AUDIO_RESAMPLER_OPT_CUBIC_B

#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B      "GstAudioResampler.cubic-b"

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


GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_B

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


GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_B

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


GST_AUDIO_RESAMPLER_OPT_CUBIC_C

#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C      "GstAudioResampler.cubic-c"

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


GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_C

G_TYPE_DOUBLE, C parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 0.0 is the default.

See GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values


GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_C

G_TYPE_DOUBLE, C parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 0.0 is the default.

See GstAudio.AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values


GST_AUDIO_RESAMPLER_OPT_CUTOFF

#define GST_AUDIO_RESAMPLER_OPT_CUTOFF      "GstAudioResampler.cutoff"

G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_CUTOFF

G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_CUTOFF

G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"

GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated. GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated. GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be interpolated. GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.


GST_AUDIO_RESAMPLER_OPT_FILTER_MODE

#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE      "GstAudioResampler.filter-mode"

GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed. GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_MODE

GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed. GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_MODE

GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed. GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"

G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables. 1048576 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables. 1048576 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables. 1048576 is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

#define GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE "GstAudioResampler.filter-oversample"

G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.


GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

#define GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR "GstAudioResampler.max-phase-error"

G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates. 0.1 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates. 0.1 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates. 0.1 is the default.


GST_AUDIO_RESAMPLER_OPT_N_TAPS

#define GST_AUDIO_RESAMPLER_OPT_N_TAPS      "GstAudioResampler.n-taps"

G_TYPE_INT: the number of taps to use for the filter. 0 is the default and selects the taps automatically.


GstAudio.AUDIO_RESAMPLER_OPT_N_TAPS

G_TYPE_INT: the number of taps to use for the filter. 0 is the default and selects the taps automatically.


GstAudio.AUDIO_RESAMPLER_OPT_N_TAPS

G_TYPE_INT: the number of taps to use for the filter. 0 is the default and selects the taps automatically.


GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION

#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"

G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.


GstAudio.AUDIO_RESAMPLER_OPT_STOP_ATTENUATION

G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.


GstAudio.AUDIO_RESAMPLER_OPT_STOP_ATTENUATION

G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.


GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH

#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"

G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH

G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.


GstAudio.AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH

G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.


GST_AUDIO_RESAMPLER_QUALITY_DEFAULT

#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4

GstAudio.AUDIO_RESAMPLER_QUALITY_DEFAULT


GstAudio.AUDIO_RESAMPLER_QUALITY_DEFAULT


GST_AUDIO_RESAMPLER_QUALITY_MAX

#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10

GstAudio.AUDIO_RESAMPLER_QUALITY_MAX


GstAudio.AUDIO_RESAMPLER_QUALITY_MAX


GST_AUDIO_RESAMPLER_QUALITY_MIN

#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0

GstAudio.AUDIO_RESAMPLER_QUALITY_MIN


GstAudio.AUDIO_RESAMPLER_QUALITY_MIN


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