GstAudioResampler
GstAudioResampler is a structure which holds the information required to perform various kinds of resampling filtering.
Methods
gst_audio_resampler_free
gst_audio_resampler_free (GstAudioResampler * resampler)
Free a previously allocated GstAudioResampler resampler.
Parameters:
resampler
–
GstAudio.AudioResampler.prototype.free
function GstAudio.AudioResampler.prototype.free(): {
// javascript wrapper for 'gst_audio_resampler_free'
}
Free a previously allocated GstAudio.AudioResampler resampler.
Parameters:
GstAudio.AudioResampler.free
def GstAudio.AudioResampler.free (self):
#python wrapper for 'gst_audio_resampler_free'
Free a previously allocated GstAudio.AudioResampler resampler.
Parameters:
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.
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:
number of input frames
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:
number of input frames
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
–
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:
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:
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.
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:
number of input frames
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:
number of input frames
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
–
in
–
input samples
in_frames
–
number of input frames
out
–
output samples
out_frames
–
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:
input samples
number of input frames
output samples
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:
input samples
number of input frames
output samples
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
–
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:
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:
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
–
in_rate
–
new input rate
out_rate
–
new output rate
options
–
new options or NULL
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.
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.
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:
method
–
flags
–
format
–
the GstAudioFormat
channels
–
the number of channels
in_rate
–
input rate
out_rate
–
output rate
options
–
extra options
The new GstAudioResampler.
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:
the number of channels
input rate
output rate
extra options
The new GstAudio.AudioResampler.
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:
the number of channels
input rate
output rate
extra options
The new GstAudio.AudioResampler.
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:
method
–
quality
–
the quality
in_rate
–
the input rate
out_rate
–
the output rate
options
–
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:
the quality
the input rate
the output rate
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:
the quality
the input rate
the output rate
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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.
Since : 1.10
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
Since : 1.10
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
Since : 1.10
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
Since : 1.10
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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