Re: laplace tranform convert to code

Started by Tim Wescott in comp.dsp15 years ago

Jerry Avins wrote: > Tim Wescott wrote: > > > tim w wrote: > > > ... > > > > -Also, the variable "fc" is the corner...

Jerry Avins wrote: > Tim Wescott wrote: > > > tim w wrote: > > > ... > > > > -Also, the variable "fc" is the corner frequency. What is that on how do > > > I set it??? > > > > In a band stop filter fc is the center frequency of the stop band. In > > your transfer function everything is scaled in time, with "To". For > > fc in Hz, fc = 1/(2 * pi * To). > > > Other tim, > > R


Re: laplace tranform convert to code

Started by Peter Nachtwey in comp.dsp15 years ago

> tim w wrote: > > Hello all, > > > > I need some guidance in programming a laplace transfer function into > > computer language --...

> tim w wrote: > > Hello all, > > > > I need some guidance in programming a laplace transfer function into > > computer language -- pseudocode for now. > > > > The transfer function is a second order function: > > > > To^2*s^2 + zeta1*To*s + 1 > > ------------------------- > > To^2*s^2 + zeta2*To*s + 1 > > That is similar to a notch filter. I posted a link to a notch filter desi


mixing a sinewave with time delayed replica of itself

Started by JohnReno in comp.dsp15 years ago 3 replies

Hello, The mathematics of frequency mixing seems so simple, but I always end up getting confused when I am trying to relate the math to what I...

Hello, The mathematics of frequency mixing seems so simple, but I always end up getting confused when I am trying to relate the math to what I observe in the lab - I don't get the correct products - have to change a sign or something to get things to work out correctly. Right now, I am trying to come up with the power transfer function of sinewave that is delayed and frequency, mixed with itself....


mixing a sinewave with time delayed replica of itself

Started by JohnReno in comp.dsp15 years ago 1 reply

Hello, The mathematics of frequency mixing seems so simple, but I always end up getting confused when I am trying to relate the math to what I...

Hello, The mathematics of frequency mixing seems so simple, but I always end up getting confused when I am trying to relate the math to what I observe in the lab - I don't get the correct products - have to change a sign or something to get things to work out correctly. Right now, I am trying to come up with the power transfer function of sinewave that is delayed and frequency, mixed with itself....


reconstruction of the input signal in time domain

Started by Pi in comp.dsp15 years ago 15 replies

What is the basic approach for the following DSP tasks: 1. Estimation of the inverse transfer function, which transform output signal to the...

What is the basic approach for the following DSP tasks: 1. Estimation of the inverse transfer function, which transform output signal to the input signal of the black-box system via known measurements of the input and output signals for both amplitude and phase characteristics simultaneously. 2. Transformation of the previously estimated inverse transfer function from spectral domain...


PreWraping in Lead/Lag

Started by Toro in comp.dsp15 years ago 6 replies

Hello I need to convert a lead/lag filter to a z transform function. I'll be using the bilinear method. Question, what equation do I...

Hello I need to convert a lead/lag filter to a z transform function. I'll be using the bilinear method. Question, what equation do I use for prewraping the lead and lag times?? For the other filters i've converted i was prewrapping the frequency and using rads/sec. For the lead/lag filters I only have a time constant input. Transfer function is T1*s + 1 g(s) ----------...


AM demodulation using MATLAB

Started by fara...@gmail.com in comp.dsp15 years ago 2 replies

Hey all, I am having some trouble with AM demodulation using MATLAB. I used the built-in MATLAB function butter to design a low pass filter and...

Hey all, I am having some trouble with AM demodulation using MATLAB. I used the built-in MATLAB function butter to design a low pass filter and using the transfer function obtained, I filtered the modulated signal multiplied by the carrier. Here's the part of my m-file which deals with it: dam=cos(2*pi*250*t).*am; % am=amplitude modulated signal [b,a]=butter(5,250*2*Ts); %designing low pa...


Partial Fraction Decomposition in c++

Started by dilpreet06 in comp.dsp15 years ago 6 replies

Hello! In order to calculate the Amplitude of a pole of an ARMA filter, I realise that I have to do a partial fraction decomposition (PFD) of...

Hello! In order to calculate the Amplitude of a pole of an ARMA filter, I realise that I have to do a partial fraction decomposition (PFD) of the ARMA filter transfer function. The PFD has to be done, so that I can calculate the residuum of the transfer function at each pole. I have the transfer function and have calculated the roots of the numerator and denominator polynomials, but have ...


linear/non-linear system

Started by VijaKhara in comp.dsp15 years ago 19 replies

hi all, 1. For a system, if the input function is u1(t), the output is y1(t) and if the input u2(t) the output y2(t). They ask if this system...

hi all, 1. For a system, if the input function is u1(t), the output is y1(t) and if the input u2(t) the output y2(t). They ask if this system is linear or not. I simply look for the transfer function: H1(s)=Y1(s)/U1(s) and H2(s)=Y2(s)/U2(s). If H1(s)=H2(s) the system is linear, if not it nonlinear. But I am confused. What if the system is linear but not time invariant. Does the proof above...


calculation of transfer function

Started by stereo in comp.dsp15 years ago 9 replies

Hi everyone, this questions sounds possibly too simple, but nevertheless I don't get it: taken a measurement, say sweep measurement of a room,...

Hi everyone, this questions sounds possibly too simple, but nevertheless I don't get it: taken a measurement, say sweep measurement of a room, I want to get the acoustic room transfer function. I have the stimulus signal X(k) and the measured response Y(k), k being the discrete frequency. All formulae I found say: the transfer function is calulated by the cross spectrum X(k)*Y(k) divided b...


transfer function

Started by VijaKhara in comp.dsp15 years ago

hi all, is the tranfer function H(s)=Output(s)/Input(s) only when the initial state of the system is zero? If the initial state of the system...

hi all, is the tranfer function H(s)=Output(s)/Input(s) only when the initial state of the system is zero? If the initial state of the system is non-zero then the Output(s)=Something*x(0)+H(s)*Input(s). where x(0) is the initial state of the system, ..(s) is the Laplace transform. Is it correct? Thanks


convert elliptic transfer coefficients to polynomial

Started by Phil Newman in comp.dsp15 years ago 1 reply

Hi, I'm trying to convert an elliptic transfer function where i have knowledge of the poles and zeros (transmission and reflection) into...

Hi, I'm trying to convert an elliptic transfer function where i have knowledge of the poles and zeros (transmission and reflection) into the s-coefficients. is there a matlab routine in which i can do this? my data is this: ========= Prototype Pole-Zero Data ========= (a + jb) S21 Poles S21 Zeros S11 Zeros -0.1805 0.89515 0 2.02 ...


Model delays in z-domain

Started by Jamake in comp.dsp15 years ago 1 reply

What would be the best way to describe the following in using discrete-time Z transfer function: A software loop filter computes a set of...

What would be the best way to describe the following in using discrete-time Z transfer function: A software loop filter computes a set of metrics using a block of data of 1024 samples (4 samples spans a symbol period). The computed metrics are then applied to the input data samples starting from next block, and so on and so forth. If I want to model this behavior in Z-domain, what would be...


impulse response computing

Started by ToF in comp.dsp15 years ago 10 replies

Hello, I'm following a course of digital image processing and a mathematical homework is first given. I need a little help for solving this...

Hello, I'm following a course of digital image processing and a mathematical homework is first given. I need a little help for solving this problem: What is the impulse response of the system: 1/ if its transfer function is G(z1,z2) =3D a + b.z1^(-1) + c.z2^(-1) + d.z1^(-1).z2^(-1) + e.z1.z2^(-1) 2/ if its frequency response if G(f,g) =3D a + b.cos(2*Pi*f) + c=2Ecos(2*Pi*g) A...


FIR filter with a flat transfer function

Started by ecco in comp.dsp14 years ago 39 replies

Hi, All While I'm not too familiar with all of the real-world DSP tricks and techniques, I do have a pretty good understanding of continuous-...

Hi, All While I'm not too familiar with all of the real-world DSP tricks and techniques, I do have a pretty good understanding of continuous- and discrete-time signals and systems. My question: I am trying to build an FIR filter for linear prediction, i.e., the 0th order tap in the FIR filter must be 0. The filter needs to have a near-flat magnitude response (1dB ripple?), and, here's ...


Transfer function of human head

Started by I. R. Khan in comp.dsp14 years ago 11 replies

Hi all, We are trying to make a hearing aid for profoundly deaf people. It is observed that if a sound signal modulated on an ultrasonic...

Hi all, We are trying to make a hearing aid for profoundly deaf people. It is observed that if a sound signal modulated on an ultrasonic carrier is conducted through the bones in the human head, it is audible (and even understood in some cases) by profoundly deaf people. However, the exact phenomenon happening inside the head is not exactly known. We want to calculate the transfer fun...


Problem of SOS matrix to Transfer Function in IIR Filter Design using Matlab

Started by X.Y. in comp.dsp14 years ago 3 replies

Hi, everyone, I have to design a IIR filter using FDATOOL in Matlab. And I expert it to workspace as SOS Matrix(SOS) and Scale Values(G). Then I...

Hi, everyone, I have to design a IIR filter using FDATOOL in Matlab. And I expert it to workspace as SOS Matrix(SOS) and Scale Values(G). Then I use function [b,a] = sos2tf(SOS,G) to acquire the Transfer Function. However, it always report error: > > [b,a]=sos2tf(SOS,G) ??? Error using ==> times Matrix dimensions must agree. Error in ==> sos2tf at 52 b = b.*g; I don't know how to handl


Loop Filter Modelling

Started by pruthvisri02 in comp.dsp14 years ago 6 replies

Hallo, I am relatively new to RF electronics,I need some support for modelling my blocks in PLL.The block we have problem is third...

Hallo, I am relatively new to RF electronics,I need some support for modelling my blocks in PLL.The block we have problem is third order LoopFilter (L.F). Here we go ---> The Transfer Function (T.F) for L.F -> (ds+1)/s(as**2+bs+c) we deduced the above T.F -> k1/s + k2/(s-r1)+k3(s-r2) where r1 and r2 are the roots of the quadratic equation in T.F & a,b,c,d are constants. And I reduced it to ti


Properties of transfer function, linear systems

Started by Peter in comp.dsp14 years ago 9 replies

Hello, Consider a linear transfer function with N-1 real and negative poles (p_n) and N real and negative zeros (z_n) factorized in the...

Hello, Consider a linear transfer function with N-1 real and negative poles (p_n) and N real and negative zeros (z_n) factorized in the standard way: (z-z_1)*(z-z_2)*...*(z-z_N) Q(z) = K ----------------------------- (z-p_1)*(z-p_2)*...*(z-p_{N-1}), where lim z-> infinity => Q(z)-> infinity. Are there then any theorems regarding such a transfer function? I am looking for pro


Transfer Function and Coherence

Started by kool_ajith in comp.dsp14 years ago 3 replies

Hi All, I have two waveforms X and Y. X is input waveform and Y is output. I have FFT algorithm with me. Now can anyone help me calculate...

Hi All, I have two waveforms X and Y. X is input waveform and Y is output. I have FFT algorithm with me. Now can anyone help me calculate the Transfer function and Coherence using the FFT, rather than Matlab etc. I have read at many places that the transfer function is calculated by the cross spectrum X(k)*Y(k) divided by the autospectrum of the input signal X(k)*X(k). But how shall...