ResampAudio [options] AFileI AFileO
Resample data from an audio file
This program resamples data from an audio file. This process involves interpolating between the samples in the original file to create a new sequence of samples with a new spacing (sampling rate). This program allows for an arbitrary ratio of output sampling rate to input rate. The default settings give a high quality sample rate conversion.
The process used for interpolation depends on the ratio of the output sampling rate to the input sampling rate.
The default interpolation filter is a linear phase FIR filter designed by applying a Kaiser window to an ideal lowpass filter response. The filter is characterized by a cutoff frequency, a window shape parameter, and the number of coefficients. The window shape parameter (alpha) controls the passband ripple and the stopband attenuation. For a fixed number of coefficients, decreasing ripple and increasing attenuation (larger alpha) come at the expense of a wider transition width.
The cutoff of the default interpolation filter depends on the input and output sampling rates. Let fsi be the sampling rate of the input signal and fso be the sampling rate of the output signal.
The default design aims for an 80 dB stopband attenuation and a transition width which is 15% of the cutoff frequency. The attenuation directly determines alpha. The value of alpha together with the transition width determines the number of filter coefficients.
The parameters of the interpolating filter can also be set by the user. The design parameters are the interpolation factor, the filter cutoff frequency, the Kaiser window parameter, and the number of filter coefficients. The following table shows the effect of changing the Kaiser window parameter alpha.
stopband alpha transition passband attenuation width D ripple 30 dB 2.210 1.536 +/- 0.270 dB 40 dB 3.384 2.228 +/- 0.0864 dB 50 dB 4.538 2.926 +/- 0.0274 dB 60 dB 5.658 3.621 +/- 0.00868 dB 70 dB 6.764 4.317 +/- 0.00275 dB 80 dB 7.865 5.015 +/- 0.00089 dB (default) 90 dB 8.960 5.712 +/- 0.00027 dB 100 dB 10.056 6.408 +/- 0.00009 dB
The filter transition width parameter is D = (N-1) dF, where N is the number of filter coefficients and dF is the normalized transition width of the filter.
Consider interpolating from 8 kHz to 44.1 kHz. The interpolation ratio is 441/80. For this example we will design the filter for an oversampling ratio of 10. This means that the filter will will be operating at a sampling rate of 80 kHz. (The default filter for this program would have used an oversampling ratio of 24.) The cutoff of the filter will be 4 kHz. The stopband attenuation is to be 80 dB. The attenuation requirement gives alpha=7.865. The parameter D corresponding to this value of alpha is 5.015. A transition width which is 15% of the cutoff corresponds to a width of 600 Hz. The normalized filter transition width of dF = 600/80000 = 0.0075. Solving for D = (N-1) dF for the number of coefficients N, gives N = 670. It is common to choose N to be of the form 2*Ir*M+1, where Ir is the filter interpolation factor (here 10). Such a time response has M sidelobes on either side of the reference point. In this example, we can choose M = 34, giving N = 681 coefficients.
If we designate the interpolation factor for the interpolation filter as Ir, about 1/Ir of the coefficients are used to calculate each output sample. The number of coefficients needed for a given value of alpha and given transition width is proportional to Ir. Increasing Ir improves the accuracy of the linear interpolation step and increases the total number of filter coefficients, but does not increase the computation effort time for the filtering operation.
For the transition width expressed as a percentage of the cutoff frequency, the number of coefficients needed to calculate each output sample is approximately 2D/P where P is the fractional bandwidth (e.g. 0.15 for a 15% transition width). The number of coefficients (rounded up) used to calculate each interpolated point is shown in the following table.
stopband alpha transition no. coeffs per output attenuation width D 15% trans. 25% trans. 30 dB 2.210 1.536 22 14 40 dB 3.384 2.228 31 19 50 dB 4.538 2.926 41 25 60 dB 5.658 3.621 50 30 70 dB 6.764 4.317 59 36 80 dB 7.865 5.015 68 42 (default) 90 dB 8.960 5.712 78 47 100 dB 10.056 6.408 87 53
On a Windows PC (1 GHz processor), with the default filter (15% transition width), this program generates about 900,000 output samples per second for general interpolation and about twice that number for ordinary interpolation.
The accuracy of the sample rate operation depends on the frequency content of the input signal. Consider increasing the sampling rate for a speech file with a 8000 Hz sampling rate. The spectrum of the signal repeats every 8000 Hz. The default filter uses a cutoff frequency of 4000 Hz with a transition width of 600 Hz. The filter passband extends to 3700 Hz and the stopband starts at 4300 Hz. The interpolation will be imperfect in that (1) frequencies falling in the lower part of the transition band will be attenuated and (2) frequencies falling in the upper part of the transition band (the image frequencies due to the repetition of the frequency response) will be only be partially attenuated. If the input signal has little energy above 3700 Hz, then the error due to both effects will be small. Tests with speech files indicate that the signal-to-distortion ratios after interpolation (say from 8000 Hz to 8001 Hz) range from 46 to 77 dB. The poorest SDR occurs for signals that have significant energy above 3700 Hz. For a fixed stopband attenuation, the SDR can be improved by increasing the number of filter coefficients to affect a decrease in the transition band width. However, the number of coefficients should not be too large, since filters with a large time span can introduce pre-echo effects.
The interpolation filter can also be read in as a filter file. For such a filter, the filter interpolation factor must be specified.
The output sample positions are determined by the output sampling rate and a sample offset parameter. The sample offset determines the position of the first output sample relative to the input samples. The default is that the first output sample coincides with the first input sample. The number of samples in the output file can also be specified. The default is to make the time corresponding to the end of the output (rounded to the nearest sample) be the same as the time corresponding to the end of the input.
".au" - AU audio file ".wav" - WAVE file ".aif" - AIFF sound file ".afc" - AIFF-C sound file ".raw" - Headerless file (native byte order) ".txt" - Text audio file (with header)
Filter file: file="file_name" Input filter file name. If specified, the filter coefficients are read from the named file. ratio=Ir Filter interpolation factor delay=Del Filter delay in units of filter samples (default for symmetrical filters is (N-1)/2, where N is the number of coefficients). The delay can be specified as a single number or as a ratio. The filter delay must be supplied for non-symmetrical filters. Windowed lowpass: ratio=Ir Filter interpolation factor. The default depends on the ratio of output sampling frequency to input sampling frequency. This parameter can be specified as a single number or as a ratio. cutoff=Fc Filter cutoff in normalized frequency relative to the filter interpolation factor (0 to Ir/2). This value can be specified as a single number or as a ratio. The default cutoff frequency is determined from the input and output sampling rates. For an increase in sampling rate, it is set to 0.5. For a decrease in sampling rate it is set to 0.5*fso/fsi. atten=A Filter stopband attenuation in dB. The attenuation must be at least 21 dB. The default is 80. The attenuation is an alternate way to specify the Kaiser window parameter alpha. alpha=a Kaiser window parameter. Zero corresponds to a rectangular window (stopband attenuation 21 dB). The default is 7.865 corresponding to a stopband attenuation of 80 dB. N=Ncof Number of filter coefficients. The default is to choose the number of coefficients to give a transition band which is 15% of the cutoff frequency. span=Wspan Window span. This is the span of the non-zero part of the window. The default window span is equal to the number of filter coefficients minus one. offset=Woffs Window offset in units of filter samples. This is the offset of the first filter sample from the beginning of the window. The default is a fractional value determined from the fractional part of the input sample offset value. gain=g Passband gain. The default gain is equal to the filter interpolation factor. This choice reproduces signals within the passband with the correct amplitude. write="file_name" Output filter file name. If specified, the filter coefficients are written to the named file.
"AU" or "au" - AU audio file "WAVE" or "wave" - WAVE file. Whether or not to use the WAVE file extensible format is automatically determined. "WAVE-EX" or "wave-ex" - WAVE file. Use the WAVE file extensible format. "WAVE-NOEX" or "wave-noex" - WAVE file; do not use the WAVE file extensible format "AIFF-C" or "aiff-c" - AIFF-C sound file "AIFF-C/sowt" or "aiff-c/sowt" - AIFF-C (byte-swapped data) "AIFF" or "aiff" - AIFF sound file "noheader" or "noheader-native" - Headerless file (native byte order) "noheader-swap" - Headerless file (byte swapped) "noheader-big-endian" - Headerless file (big-endian byte order) "noheader-little-endian" - Headerless file (little-endian byte order) "text-audio" - Text audio file (with header)
See routine CopyAudio for a description of other options.
ResampAudio -i 1 abc.au new.au
ResampAudio -i 1 -a -1/8 abc.au new.au
ResampAudio -s 8001 abc.au new.au
ResampAudio -i 6 abc.au new.au
This environment variable specifies a list of directories to be searched when opening the input audio files. Directories in the list are separated by colons (semicolons for Windows).
P. Kabal / v10r5 2023-04-10