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DSP Library Functions list:
dspl_bessel_i0(1)  (x) modified bessel function of the first kind I0(x). dspl_butter_ap(2)  (rp,ord) calculates the transfer function H(s) coefficients of analog normalized lowpass Butterworth filter. dspl_butter_ap_zp(2)  (ord,rp) calculates arrays of zeros and poles for analog normlized lowpass Batterworth filter transfer function H(s). dspl_cfft(1)  (x) calculated npoints FFT for the complex vector x. dspl_cgoertzel(2)  (x,ind) calculates samples of npoint DFT, according to ind indexes vector (Goertzel algorithm). dspl_cheby1_ap(2)  (rp,ord) calculates the transfer function H(s) coefficients of analog normalized lowpass Chebyshev type 1 filter. dspl_cheby1_ap_zp(2)  (ord,rp) calculates arrays of zeros and poles for analog normlized lowpass Chebyshev type 1 filter transfer function H(s). dspl_cheby2_ap(2)  (rs,ord) calculates the transfer function H(s) coefficients of analog normalized lowpass Chebyshev type 2 filter. dspl_cheby2_ap_zp(2)  (ord,rs) calculates arrays of zeros and poles for analog normlized lowpass Chebyshev type 2 filter transfer function H(s). dspl_cidft(1)  (x) calculates the npoint inverse discrete transform Fourier complex spectrum x. dspl_cifft(1)  (x) calculates npoint IFFT of complex data x. dspl_concat(2)  ( a,b ) concatenate arrays a and b. dspl_conv(2)  ( a,b ) convolves two complex vectors. dspl_decimate(2)  (x,d) decimates the real vector x by d times. dspl_cdft(1)  (x) calculates the npoint discrete Fourier transform complex signal x. dspl_dft(1)  (x) calculates the npoint discrete Fourier transform real signal x. dspl_ellip_ap(3)  (rp,rs,ord) calculates the transfer function H(s) coefficients of analog normalized lowpass elliptic filter. dspl_ellip_ap_zp(3)  (ord,rp,rs) calculates arrays of zeros and poles for analog normlized lowpass elliptic filter transfer function H(s). dspl_fcoeff(4)  (t,x,T,nw) calculates Fourier series coefficient for real vector x. dspl_fft(1)  (x) calculated npoints FFT for the real vector x. dspl_fft_mag(2)  (x,fs) calculates the frequency response for a real vector x. dspl_fft_mag(3)  (x,fs,flag) calculates the frequency response for a real vector x. dspl_fft_shift(1)  (x) perform a shift of the vector x, for use with the fft and ifft functions, in order to move the frequency 0 to the center. dspl_filter_freq_resp(5)  (b,a,ord,w,flag) calculates magnitude, phase response and group delay vectors for digital or analog filter. dspl_filter_iir(4)  (b,a,ord,s) calculates real IIR filter output for real signal s. dspl_fir_linphase(6)  (ord,w0,w1,ftype,wtype,wprm) calculates linearphase FIR filter coefficients by window method. dspl_flipip(1)  (x) flips complex vector x in the memory. dspl_freqs(4)  (b,a,ord,w) calculates the values of the complex gain H(jw) of the analog filter. dspl_freqz(4)  (b,a,ord,w) calculates the values of the complex gain H(e^jw) of the digital filter. dspl_fseries(2)  (ws,t) reconstructs the time function from Fourier series coefficients. dspl_fseries(3)  (w,s,t) reconstructs the time function from Fourier series coefficients. dspl_goertzel(2)  (x,ind) calculates samples of npoint DFT, according to ind indexes vector (Goertzel algorithm). dspl_group_delay(5)  (b,a,ord,flag,w) calculates group delay for digital or analog filter. dspl_iir(6)  (rp,rs,ord,w0,w1,ftype) calculates the coefficients of the digital IIR filter transfer fucntion H(z). dspl_linspace(4)  (x0,x1,n,type) fills a vector with n linearly spaced elements between x0 and x1. dspl_logspace(4)  (x0,x1,n,type) fills a vector with n logarithmically spaced elements between 10^x0 and 10^x1. dspl_ones(1)  (n) returns vector filled by ones values. dspl_phase_delay(5)  (b,a,ord,flag,w) calculates phase delay for digital or analog filter. dspl_randb(1)  (n) generates a binary unipolar [0, 1] pseudorandom vector. dspl_randb2(1)  (n) generates a binary unipolar [1, 1] pseudorandom vector. dspl_randi(3)  (n,start,stop) generates a pseudorandom vector of integers ranging from start to stop inclusive. dspl_randinit(2)  (type,seed) pseudorandom numbers generators initialization. dspl_randn(3)  (n,mu,sigma) generates a vector of normally distributed pseudorandom numbers. dspl_randu(1)  (n) generates a vector of uniformly distributed pseudorandom numbers in the range from 0 to 1. dspl_scale_lin(5)  (x,xmin,xmax,dx,h) vector x linear transformation. dspl_sum(1)  (x) returns the sum of the elements of a real vector x. dspl_sum_sqr(1)  (x) returns the sum of squares of the elements of a real vector x. dspl_window(2)  (n,wtype) calculates a periodic or symmetric window function. dspl_window(3)  (n,wtype,wprm) calculates a periodic or symmetric window function.
DSPL2.0 — opensource crossplatform digital signal processing algorithm library, written in C language. Distributed under LGPL v3 license. This allows to use this library in all applications with dynamic linking. Source codes are available on GitHub. DSPL2.0 includes follow algorithms sets:  Digital spectral analysis, discrete and fast Fourier transform algorithms.
 Analog and digital IIR filters design and analysis.
 Digital FIR filters design and analysis.
 Windows function collection includes 15 different parametric and nonparametric window functions.
 Digital Hilbert transform algorithms.
 Mathematical sections includes trigonometric, hyperbolic, elliptic functions of real and complex variables.
 Pseudorandom numbers generation algorithms.
 Statistic functions.
 Linar algebra algorithms (BLAS and LAPACK packages are used under the hood of the DSPL2.0 library).
 Digital resampling algorithms.
Examples: dspl_filter_iir.pdf (156kb) downloaded 74 time(s). dspl_fir_linphase.pdf (247kb) downloaded 55 time(s). dspl_iir.pdf (234kb) downloaded 43 time(s). dspl_window.pdf (857kb) downloaded 58 time(s). Links: 1. DSPL2.0 — free DSP algorithm library . Edited by user 12 December 2021 22:38:42(UTC)
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Russia ☭ forever Viacheslav N. Mezentsev 
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