DSPRelated.com

Attempting DSP in Scilab -- filtering using convol()

Started by Richard Owlett in comp.dsp16 years ago 5 replies

I'm attempting to use convol() to implement a simple filter. Scilab's Help entry is terse to say the least. I suspect I should be using the...

I'm attempting to use convol() to implement a simple filter. Scilab's Help entry is terse to say the least. I suspect I should be using the overlap and add variant, but Scilab gives no example. I've experimented and wrote the following code. Calculating the convolution by both methods. The results can only be described as "similar". What am I missing? TIA t=[0:.00001:1]; s=si...


PhD s from Convolution Integral Thread

Started by Fred Marshall in comp.dsp14 years ago 46 replies

It got too long and I just couldn't spend the time to read all the posts. So, if this is redundant then .. OK: 1) Folks with common sense...

It got too long and I just couldn't spend the time to read all the posts. So, if this is redundant then .. OK: 1) Folks with common sense and experience are invaluable. 2) Folks with an ability to focus on the real issue are invaluable. 3) Folks who can figure something out quickly and put what they know to use are invaluable. So, if in the process of earning a degree a person becomes...


Generation of synthetic seismic trace with known Q-value

Started by Nicholas Kinar in comp.dsp14 years ago 8 replies

Hello-- I would like to generate a synthetic seismic trace using the real-valued Ricker wavelet. (Other wavelets could also be used.) ...

Hello-- I would like to generate a synthetic seismic trace using the real-valued Ricker wavelet. (Other wavelets could also be used.) Normally I think that this should involve a convolution operation. However, I would like to ensure that the synthetic trace has a constant (and known) Q-value. I would like to find a monograph/book/paper or some other type of procedure to efficientl...


challenge: how to do filtering in DCT domain?

Started by walala in comp.dsp20 years ago 1 reply

Dear all, I guess this is a little simple... Don't largh at me if it is too simple... Basicall I have a 3x3 filter, let's say [ -0.1 0.2...

Dear all, I guess this is a little simple... Don't largh at me if it is too simple... Basicall I have a 3x3 filter, let's say [ -0.1 0.2 -0.1; 0.2 0.6 0.2; -0.1 0.2 -0.1]; I want to do this in DCT domain... I know convolution in spatial domain is multiplication in frequency domain... but here the DCT ...


Time domain convolution in a real-time situation

Started by DSP-Newbie in comp.dsp17 years ago 14 replies

Hi group, I'm still working on my software BFSK-modem project. More specifically, i want to use better filters, so I'm experimenting with a...

Hi group, I'm still working on my software BFSK-modem project. More specifically, i want to use better filters, so I'm experimenting with a windowed sinc filter (Blackman window). I can succesfully calculate the impulse response H[] My problem is that when I apply the filtering algorithm ( as found in ch. 16 of The Scientist & Engineer's Guide to DSP ) , that the first M output datap...


q: Smith book code segment 3090 ch 26

Started by waltech in comp.dsp15 years ago

Neural Networks,ch 26,, p. 479 online ( yes, the hardback has already been ordered) In the following code, I can't seem to follow how the...

Neural Networks,ch 26,, p. 479 online ( yes, the hardback has already been ordered) In the following code, I can't seem to follow how the convolution is calculated. I'm pretty sure I follow the basic technique in ch6-7, but perhaps the use of the variables and the complex format make it difficult. I certainly see an empty REX and IMX, except for the "impulse" inserted at IMX[12].. I...


Which version of Autocorrelation function to use?

Started by m26k9 in comp.dsp16 years ago 4 replies

Hello, I think I am missing something very basic here. I am confused as to which definition of autocorrelation to use. From Wikipedia...

Hello, I think I am missing something very basic here. I am confused as to which definition of autocorrelation to use. From Wikipedia (http://en.wikipedia.org/wiki/Autocorrelation), there are two definitions. 1) Based on convolution 2) Based on 'expectation' - E[f(t).f*(t-t')] I am not sure which version to use. I am working on OFDM so I will be mostly interested in calculation the a...


Identifying this pseudo convolution operation

Started by L13 in comp.dsp14 years ago 3 replies

Hi all, I am having trouble identifying an input/output relationship in a communication system. In the course of my research, I encounter an...

Hi all, I am having trouble identifying an input/output relationship in a communication system. In the course of my research, I encounter an input/output relationship of this type at time n: y_n = \sum_{l} \sum{l'} a_{n,l,l'} x_{n-l-l'} (The summations are over all l,l' which are finite. It is the result of a transmission through a doubly selective channel) I am not sure what it r...


not a descrete case but a simple question about signal processing

Started by VijaKhara in comp.dsp17 years ago 1 reply

Hi all I am reading the Papoulis's book on Random Variable and getting confused with the following formula: X(t) is input of a linear...

Hi all I am reading the Papoulis's book on Random Variable and getting confused with the following formula: X(t) is input of a linear system with impulse response h(t), Y(t) is its output. X(t) is a WSS Random Process ===> the Cross Correlation: Rxy(to)= Rxx(to) * h*(-to). where * is convolution, and h*(-to) is conjugate of h(-to). Now the density spectrum Sxy(omega)= Sxx(omega)H*(


Downsampling

Started by samersamy in comp.dsp13 years ago 4 replies

Hi all, It is my first time to post here. I working on something that involves down-sampling an audio stream from 44.1 Khz to 5512 Hz. After...

Hi all, It is my first time to post here. I working on something that involves down-sampling an audio stream from 44.1 Khz to 5512 Hz. After a lot of reading i end up with implementing a Windowed-sinc low-pass-filter with a cut of frequency 2756 Hz (half 5512 Hz). After that i use FFT to make convolution. My filter Java code implementation works fine and also i followed some algorithm in "T...


Frequency response function...

Started by Atmapuri in comp.dsp19 years ago 2 replies

Hi! I have an s domain transfer function H(s) and trying to determine the response of the system to the unit step function. I tried...

Hi! I have an s domain transfer function H(s) and trying to determine the response of the system to the unit step function. I tried this: Y(s) = H(s) * X(s) = H(s) * Hc(s) = H(s)/s (* is product, not convolution) But the frequency response using freqs(Y(s)) is wrong. (I bet it must be obvious to someone why). The frequency spectrum should a have a value at DC (omega*j = 0) but ...


which FPGA chip to use for FFT?

Started by me_rythm in comp.dsp16 years ago 2 replies

Hello, I have made a program which calculate FFT of two images (1024X1024) and then convolve them. I wish to know which FPGA chips would be...

Hello, I have made a program which calculate FFT of two images (1024X1024) and then convolve them. I wish to know which FPGA chips would be best for this task, I want the result of the algorithm in real time (more than 30hz). It takes around 10 seconds on my machine(AMD 1.8 Ghz Dual Core) for just one convolution of both the images. I am thinking to use xilinx XC3S500E (500k gates). I have no e...


Deconvolution procedure by Weiss method?

Started by anthony in comp.dsp18 years ago 2 replies

In my understanding, the deconvolution process with FFT is to use divide operator instead of the deconvolution operator, that is ---aries44's...

In my understanding, the deconvolution process with FFT is to use divide operator instead of the deconvolution operator, that is ---aries44's post----------- In order to convolve two functions 'a' and 'b', we can take their Fourier Transform(FT) and multiply them in Fourier domain i.e. C= FT(a) * FT(b) c = IFT(C) and then Inverse Ft(IFT) of 'C' gives us the convolution of 'a' and 'b'. Now i...


Order of convolution and multiplying

Started by ejs in comp.dsp16 years ago 8 replies

Hi all, I have three signals in the frequency domain, which arranged in mathematical way as follow: The first signal (named as M) is...

Hi all, I have three signals in the frequency domain, which arranged in mathematical way as follow: The first signal (named as M) is multiplied with a signal (named as R) that blocks some frequency components and transmits the others, and the result is convolved with the first signal (M). My question is, what are the mathematical operations that I have to do if I need to rearrange the ord...


infer a 2D filter from its frequency spectrum

Started by Anonymous in comp.dsp16 years ago 2 replies

Dear all, I am working on a 2D convolution problem. I have estimated the frequency spectrum |H(u, v)| now I wish to infer the 2D filter...

Dear all, I am working on a 2D convolution problem. I have estimated the frequency spectrum |H(u, v)| now I wish to infer the 2D filter f(x, y). I just don't know a way to work it out from the spectrum, although I may have several constrains of f(x, y) in the spatial domain for assistance. It will also be helpful if there's a good way to easily compute derivative d|H(u, v)|/df(x, y), tha...


Matched Filter In the Frequency Domain Question

Started by westocl in comp.dsp13 years ago 5 replies

Maybe somebody can help me think about this the right way. Ok, it is well known that there is an 'equivalence' in convolution in the time...

Maybe somebody can help me think about this the right way. Ok, it is well known that there is an 'equivalence' in convolution in the time domain to multiplication in the frequncy domain, so consider this... Say we have a modulated bitstream of lets say 10 bits and knew what they were., then did a simple matched filter operation (on the modulated signal maybe perurbed by noise that corrup...


Zero padding and Cross Correlations

Started by dspchick in comp.dsp19 years ago 1 reply

Hi, Just say I am using the FFT (in matlab) to compute the crosscorrelation of 2 discrete time signals, each of length N. I've tried 2...

Hi, Just say I am using the FFT (in matlab) to compute the crosscorrelation of 2 discrete time signals, each of length N. I've tried 2 cases 1) zero-padding to a length of 2N, in order to avoid circular convolution (aliasing effects) 2) zero-padding to a length of 8N. I expected that case 2) would give me a better estimate on the lag, but it seems like the estimate is worse! Is t...


How to add noise???

Started by cpshah99 in comp.dsp16 years ago 5 replies

Dear all It seems I have forgot all my fundamentals. Can you please guide me on how to add noise to the system. So far I did simulations at...

Dear all It seems I have forgot all my fundamentals. Can you please guide me on how to add noise to the system. So far I did simulations at baseband where we just map the data to symbols and add noise. i.e. for 1/2 rate convolution and 4-QAM modulation, SNR=Eb/NO=1/(sigma^2); and from this find the sigma and multiply with random sequence. But how does this differ at passband? Where my ...


Mapping MSK ? How to regain the information from I and Q ?

Started by Florian Toulouse in comp.dsp18 years ago

Hello , I despair on how to regain the initial information from the seperate I and Q bits after MSK modulation/demodulation. If d(t) is...

Hello , I despair on how to regain the initial information from the seperate I and Q bits after MSK modulation/demodulation. If d(t) is the data, a sum of delta pulses g(t) is the frequency response (MSK: REC) and c(t)=g(t)*d(t) (* convolution !) then phi(t) is the phase , which is calculated by cumsum(c(t)) (MSK:"linear ramps up and down according to the bit") an...


Audio Equalization Using MDCT

Started by foxcob in comp.dsp15 years ago 1 reply

I have done real-time audio equalization using fast-convolution (overlap-add) where the frequency domain values were specified by the end user...

I have done real-time audio equalization using fast-convolution (overlap-add) where the frequency domain values were specified by the end user through a graphical interface. If the phase response did not need to be specified, could a MDCT give better frequency resolution for around the same number of computations? Would this work or am I completely off here. Thanks, Jacob