transfer function

Started by Anonymous in comp.dsp14 years ago 14 replies

I have a transfer function as follows: K*z^(-10) ---------------------- 1 - z^(-1) + K*z^(-10) I'm trying to figure out for what...

I have a transfer function as follows: K*z^(-10) ---------------------- 1 - z^(-1) + K*z^(-10) I'm trying to figure out for what values of K the system is stable. Is there an easy way to do this? Thanks. BTW, this isn't homework.


resonant frequency

Started by mike...@gmail.com in comp.dsp14 years ago 4 replies

I have a transfer function that is: H(z) = Kz^(-T) ----------------------- 1 - z^(-1) + K*z^(-T) T is a time...

I have a transfer function that is: H(z) = Kz^(-T) ----------------------- 1 - z^(-1) + K*z^(-T) T is a time delay K is a loop gain factor I was wondering if there is a way to calculate what the resonant frequency of this system is. Any help is appreciated. Thank you.


Adaptive Notch Filter - LMS Algorithm

Started by zrimkunas in comp.dsp14 years ago 4 replies

Hi everyone, I am working on a simple adaptive notch filter that will be used to cancel one sinusoid. Additionally, the filter coefficient is...

Hi everyone, I am working on a simple adaptive notch filter that will be used to cancel one sinusoid. Additionally, the filter coefficient is complex (I am looking after the error envelope). So what I have is one zero and one pole. The zero is fixed to the unit circle and the pole is very close to the unit circle at the same angle as the zero. The transfer function is: H(z) = (1 - (1


Transfer function and Coherence using Matlab

Started by kool_ajith in comp.dsp14 years ago 4 replies

Hi All, 1) I have a windows application created using .NET, C# to be specific. 2) I have two waveforms X and Y. X is input waveform and...

Hi All, 1) I have a windows application created using .NET, C# to be specific. 2) I have two waveforms X and Y. X is input waveform and Y is output. X and Y are nothing but 1048 samples of complex data in an Array. 3) I have FFT algorithm with me. 4) Now can anyone help me calculate the Transfer function and Coherence using Matlab. I think Matlab has some function call to calculate...


Convert Analog Transfer Function to Digital?

Started by Chris Barrett in comp.dsp14 years ago 2 replies

Let's say I have the analog transfer function b(1)s^n + b(2)s^(n-1) + b(3)s^(n-2) H(s) = -------------------------------------- ...

Let's say I have the analog transfer function b(1)s^n + b(2)s^(n-1) + b(3)s^(n-2) H(s) = -------------------------------------- a(1)s^m + a(2)s^(m-1) + a(3)s^(m-2) Can I convert this to a digital transfer function in a straight forward manner? Would the digital equivalent be b(1) + b(2)z^-1 + b(3)z^(-2) H(z) = -------------------------------- ...


Numerically Computing Impulse Response from Power Frequency Response

Started by sush...@yahoo.com in comp.dsp14 years ago 5 replies

I am trying to compute the impulse response of a given channel to include in a BER analysis. I have the power transfer function of the channel,...

I am trying to compute the impulse response of a given channel to include in a BER analysis. I have the power transfer function of the channel, not the classical amplitude/phase transfer function of the channel. Fortunately, the transfer function is symmetric, so I can numerically calculate the impulse response as: amp_freq_resp = sqrt(power_freq_resp); imp_resp = ifft(amp_freq_resp); i...


All-Pass filter

Started by Anonymous in comp.dsp14 years ago 3 replies

If I have a digital all-pass transfer function (IIR) and I divide out the numerator/Denominator - is there a theorem somewhere which says that...

If I have a digital all-pass transfer function (IIR) and I divide out the numerator/Denominator - is there a theorem somewhere which says that the 'infinite' polynomial I get has few higher order coefficients that are significant? ie the higher order terms tend towards zero..?? suppose I have (1-2z^-1)/(1-0.5z^-1) for instance or a second-order example.. Thanks W.K


Accuracy when Setting Center Frequency

Started by Chris Barrett in comp.dsp14 years ago 7 replies

When one discretizes a transfer function, does one loose accuracy in the center frequency? If yes, how can one compensate so that accuracy is...

When one discretizes a transfer function, does one loose accuracy in the center frequency? If yes, how can one compensate so that accuracy is not lost? I have an analog transfer function. When ever I run the bilinear transform on it and plot the frequency response, the center frequency is shifted.


Differentiating impulse response

Started by gkn in comp.dsp14 years ago 10 replies

As differentiation in the time domain corresponds to multiplication by s in the Laplace domain, it seems reasonable (to me) in terms of the...

As differentiation in the time domain corresponds to multiplication by s in the Laplace domain, it seems reasonable (to me) in terms of the Bilinear transform that multiplying the same digitized transfer function by 1-z^?? ------ 1+z^?? should give the same desired effect in discrete case. But why doesn't this work? (If I, however, just multiply by 1-z?? it ofcourse corresponds to a finit...


How to design FIR filter with matematical equation?

Started by c1910 in comp.dsp14 years ago 1 reply

hi, i'm very new with DSP FIR. i need to make a Bandpass filter with FIR design. first, i use the AM signal for the input, then i want to get...

hi, i'm very new with DSP FIR. i need to make a Bandpass filter with FIR design. first, i use the AM signal for the input, then i want to get the information signal by filtering the AM signal. The problem is to get the value of transfer function of the FIR filter, i need a mathematical equation, i don't know where to start and what equation or theory to use?! can you give me an equation to ...


Re: How to design FIR filter with matematical equation?

Started by Clay in comp.dsp14 years ago

On May 23, 8:11 am, "c1910" wrote: > hi, > i'm very new with DSP FIR. > i need to make a Bandpass filter with FIR design. first, i use...

On May 23, 8:11 am, "c1910" wrote: > hi, > i'm very new with DSP FIR. > i need to make a Bandpass filter with FIR design. first, i use the AM > signal for the input, then i want to get the information signal by > filtering the AM signal. > The problem is to get the value of transfer function of the FIR filter, i > need a mathematical equation, > i don't know where to


Re: How to design FIR filter with matematical equation?

Started by Clay in comp.dsp14 years ago

On May 23, 8:11 am, "c1910" wrote: > hi, > i'm very new with DSP FIR. > i need to make a Bandpass filter with FIR design. first, i use...

On May 23, 8:11 am, "c1910" wrote: > hi, > i'm very new with DSP FIR. > i need to make a Bandpass filter with FIR design. first, i use the AM > signal for the input, then i want to get the information signal by > filtering the AM signal. > The problem is to get the value of transfer function of the FIR filter, i > need a mathematical equation, > i don't know where to


thx for clay and julius, 'Still mathetical problem, but for AM demodulation with FIR filter"

Started by c1910 in comp.dsp14 years ago 9 replies

Clay, thanks for your opinion... my problem is the mathematical equation for how the FIR filter to process demodulation for AM. oya, bout the...

Clay, thanks for your opinion... my problem is the mathematical equation for how the FIR filter to process demodulation for AM. oya, bout the windowing method, u tell me...i found the equation for bandpass filter ideal. i read a book, and i got the equation for unideal filter. is it h=w.hd? h:transfer function unideal w:windowing coef. hd: transfer function ideal filter and do i hav...


Warping Pole/Zero Plot

Started by Chris Barrett in comp.dsp14 years ago 4 replies

I'm converting between to s domain and the z domain by finding the poles and zeros. My plots of the frequency response look correct, but the...

I'm converting between to s domain and the z domain by finding the poles and zeros. My plots of the frequency response look correct, but the scaling on the y-axis is not the same. Does any one know how I fix this? Here are some more details: What I have is a continuous time transfer function given by 2*w_d h(s) = --------------------- s^2 ...


Bob Carver technology at guitar pickups

Started by MustafaUmut in comp.dsp14 years ago 11 replies

As you know Bob Carver copied a 25000 dollars worth of amp with 500 dollars worth of circuit. He copied the transfer function of the expensive amp...

As you know Bob Carver copied a 25000 dollars worth of amp with 500 dollars worth of circuit. He copied the transfer function of the expensive amp with the cheap one. No body could not understand the difference between two amps. Honestly talking , I am not a scientist but want to copy Fender Jazz Bass pickup with this method. Can it be done and how ? Guitar pickups are made from a magnet and wir...


minimum phase HRTF

Started by Emile in comp.dsp14 years ago 2 replies

Hello, i'm looking for a tool or matlab function to convert head related transfer functions (impulse responses) into a minimum phase filter +...

Hello, i'm looking for a tool or matlab function to convert head related transfer functions (impulse responses) into a minimum phase filter + corresponding allpass filter (or phase excess in the form of nr of samples for the interaural time delay). I would also be happy with a dataset that already has been converted. I've read through the minimum phase chapter in the oppenheim&schafer ...


Calculating filter coefficients from magnitude squared filter

Started by Anonymous in comp.dsp14 years ago 2 replies

Good Morning, As part of my current research, I am looking into applying IIR filters to generate noise with a variety of spectra. These...

Good Morning, As part of my current research, I am looking into applying IIR filters to generate noise with a variety of spectra. These filters need to be designed at runtime. Only the magnitude of the filter's transfer function is relevant - the phase of the white noise process used to drive to filter is already random. Because of this, I can use one of the many magnitude-squared filter d...


ANC Headphone

Started by pooyad in comp.dsp14 years ago 2 replies

Hi, I'm working on ANC headphone. Because I could'nt implement it with dsp board (here in iran these boards are unavailable) I need Primary and...

Hi, I'm working on ANC headphone. Because I could'nt implement it with dsp board (here in iran these boards are unavailable) I need Primary and Secondary Paths transfer functions before starting simulating it in matlab. Would you please help me, I need discrete-time model of these transfer functions for specific headphone.


Help With Interpretation of Pole Locations in Z-Plane

Started by Greg Berchin in comp.dsp14 years ago 4 replies

I am modeling a system using a conventional linear predictor, in which the present output sample is predicted as a linear combination of...

I am modeling a system using a conventional linear predictor, in which the present output sample is predicted as a linear combination of past samples. Having obtained the transfer function of the analysis filter from those prediction coefficients, I am formulating the synthesis filter from the inverse of the analysis filter. All standard stuff, and my question is not about any aspect of th...


Filter to Immitate Gramophone

Started by panabiker in comp.dsp14 years ago 14 replies

I wonder if there exist some spec or transfer function of a filter that would immitate the responce of an old gramophone or 78s. I...

I wonder if there exist some spec or transfer function of a filter that would immitate the responce of an old gramophone or 78s. I am retrfitting an old gramophone with modern mp3 electronics but like it to sound more "authentic".