Hello, I'm reading a lot of literature on windowing in spectral analysis but haven't found theoretical cause of background for statement that...

Hello, I'm reading a lot of literature on windowing in spectral analysis but haven't found theoretical cause of background for statement that windows sequences should be nonnegative. It seems logical that weighting should be done with positive factors but since gained frequency response is convolution integral between "real" response and window's response there seems to be no cause for no...

I have been trying to develop some microphone modelling software that uses fast convolution with impulse responses to apply the frequency...

I have been trying to develop some microphone modelling software that uses fast convolution with impulse responses to apply the frequency response of a mic to a sound. So far i have got this working fine. However, I want to be able to remove the frequency response of the source microphone before applying the target microphone model. I'm sure this should be possible using the impulse res...

I've been coding up a working MATLAB simulation on a Blackfin 533 using the 32 bit fractional data type (fract32). I have used FIR filters...

I've been coding up a working MATLAB simulation on a Blackfin 533 using the 32 bit fractional data type (fract32). I have used FIR filters to implement convolution to get outputs from different filter stages. I'm using look up tables created from MATLAB to simulate the algorithm internally on the processor. The LUTs consist of secondary path filtered white noise and a 30Hz sine wave also filtered...

The basic question is how would I know the Frequency Reponse of a "black box" system if all I had was the inputs going in and output samples...

The basic question is how would I know the Frequency Reponse of a "black box" system if all I had was the inputs going in and output samples coming out? I know I could use convolution to get a best-fit impulse response, but given these coefficients how would I calculate a frequency response? I guess I'm almost wondering how Matlab does it in the freqz() function?! I found this thread that a...

Hello, I was looking at the DVB-H standard which has a serially concatenated encoding with Reed Solomon code (RS(204,188, t=8) from...

Hello, I was looking at the DVB-H standard which has a serially concatenated encoding with Reed Solomon code (RS(204,188, t=8) from RS(255,239, t=8) ) as the outer code and Convolution code (K=7) as the inner code. At the receiver, I can think of at least four decoding options, 1. Viterbi followed by RS decoder. 2. MAP (BCJR) followed by RS decoder. 3. MAP followed by soft input RS decode...

I just assumed "matched filtering" included the deconvolution and other steps to recover the original signal's shape. In this case it was...

I just assumed "matched filtering" included the deconvolution and other steps to recover the original signal's shape. In this case it was actually easier to invent a new filter -- I assumed it already existed -- than to be scholarly and do some research. To avoid confusion with the terminology from now on the filter that takes the convolution of a noisy signal like a conventional matche...

Hello all, I'm trying to implement a frequency-domain filter on the TI C55x and encountered some problems with accuracy. My code takes the...

Hello all, I'm trying to implement a frequency-domain filter on the TI C55x and encountered some problems with accuracy. My code takes the FFT of the input and the coefficients, multiply them, and applies iFFT to get the time-domain output back. However, when I tested the algorithm with random data, the error (compared to Matlab convolution) is not centered at zero for the last few sampl...

Hi all. please help me with this code,I dont know what is my mistake function[y]=myconv(x,h) y=zeros(length(h),length(x)) for j=1:length(h) ...

Hi all. please help me with this code,I dont know what is my mistake function[y]=myconv(x,h) y=zeros(length(h),length(x)) for j=1:length(h) for i=1:length(x) if(j> =i) y(i,j)=x(i)*h(j); end end yn=zeros(1,length(x)+length(h)-1)

Halo, I am new to SystemC. I work on a SystemC simulation chain. I get this error when i try to run my program! , code is, if...

Halo, I am new to SystemC. I work on a SystemC simulation chain. I get this error when i try to run my program! , code is, if (TmpChannelType != AWGN) { // channel convolution ref_siso_channel_funct ( ptInSymbols, ptOutSymbols, ptCtrl); CpxOutPort2.write ( ptCtrl-> ptChannelCoeff, TmpNbCarriers) // writes the channel coeffeicients of size "TmpNbCarriers" to complex port 2

CAVEAT - My formal coursework in "Signals and Systems" predates undergrads being introduced to Cooley, Tukey etc. I'm also such an antique...

CAVEAT - My formal coursework in "Signals and Systems" predates undergrads being introduced to Cooley, Tukey etc. I'm also such an antique that I visualize in terms of RLC tank circuits characterized by Q ~= F0/BW. (My father operated a legal land based spark gap transmitter and built his own capacitors in a fish tank obtaining ~ 1 ufd/gal {no idea what the insulation was} ;) BACKGROU...

Hi, Fading channel models (and noise) are usually specified for a modulated waveform. For e.g. Jakes model can be used to generate taps and...

Hi, Fading channel models (and noise) are usually specified for a modulated waveform. For e.g. Jakes model can be used to generate taps and fade the modulated waveform. I would like to know if there is a way in which the channel can be modeled whose I/P is a binary stream of data for e.g. O/P stream of a convolution encoder. The end objective is to by-pass the modulation - dem

Hi all, In a book they have this derivation which I don't understand, Please help me out. |delta(omega)-H(omega)|^2 ----Inverse FT--->...

Hi all, In a book they have this derivation which I don't understand, Please help me out. |delta(omega)-H(omega)|^2 ----Inverse FT---> (delta(n)-h(n))**(delta(n)-h(-n)) where | | denotes the ablosulte operator, h(n) is the inverse FT of H(omega), and ** is the convolution operator. What I am confused are as follows: 1. Why there is an absolute operator in the frequency domain

Hi , I am trying to model a system ( in MATLAB ) where i have a OFDM system .so i start with generating a random set of data , convert the...

Hi , I am trying to model a system ( in MATLAB ) where i have a OFDM system .so i start with generating a random set of data , convert the data to a parallel form, then a QPSK modulation ,perform a IFFT and then insert a guard interval . now what i intend to do is pass it through a underwater channel model and check for the error rate vs SNR . so in parallel i have a channel matrix as a result...

Im an undergraduate in DSP having now completed the taught modules of the course covering the usual introduction level signal processing theory...

Im an undergraduate in DSP having now completed the taught modules of the course covering the usual introduction level signal processing theory such as sampling theorm, fourier analysis, convolution, correlation. I'm now undertaking my thesis, the detection of modern comms signals, and have been reading up on the associated wealth of theory and papers, in particular statistical signal processing. ...

Hi, I recently looked into some old image processing program to see how bump map shading was done, and found out this simple convolution...

Hi, I recently looked into some old image processing program to see how bump map shading was done, and found out this simple convolution kernel : 1 1 1 1 -2 1 -1 -1 -1 With an offset of 128, this turns a bump map into a pretty good shaded map. I'd like to use that to shade a bump mapped sphere in real time (for a demo), but to do so I'd need to be able to calculate such a kern...

Hi, Ive been trying to obtain BER curves for a wireless system with and without FEC. I understand that there will be certain amount of...

Hi, Ive been trying to obtain BER curves for a wireless system with and without FEC. I understand that there will be certain amount of coding gain depending on the rate of the code. I want to use convolution encoding, but I dont think there is a linear relationship between the rate of the code and the coding gain. Can anybody recommend a book or website where I could learn how to obtain my...

Dear friends, Here I have a problem to re-produce same results with MATLAB and FFTW during 2 dimensional FFT. I'm trying to do FFT with...

Dear friends, Here I have a problem to re-produce same results with MATLAB and FFTW during 2 dimensional FFT. I'm trying to do FFT with Gaussian convolution which coordinates varies from negative to positive, FFTW3 seems not to calculate the values correctly. It happens only near center place. And it affects IFFT and it's shifted results seriously in my calculation. ---------------------...

Folks I have been dubugging some software on a 6713 DSK using an FFT. I've isolated the problem down to either (a) running the interrupt...

Folks I have been dubugging some software on a 6713 DSK using an FFT. I've isolated the problem down to either (a) running the interrupt driven McBSP codec at more than 24kHz sampling rate or (b) running the fast convolution in the main() function using the TI DSPLIB DSPF_sp_cfftr2_dit function with an FFT size greater than 256 points. The stack dump indicates that it's overflowed, and the...

hello, i am trying to compare my versus matlab's implementaion of a convolution in the frequency domain. i have a mask that is not the...

hello, i am trying to compare my versus matlab's implementaion of a convolution in the frequency domain. i have a mask that is not the same size of the image (in fact its much smaller). but i fft2 the mask with the dimensions of the image and then convolve with the fft2 of the image. my question i, what method is used for zero padding the mask to be the same size of the image? can...

Hi everyone... I'm a newbie... I have a question. I have to make something like y = h1*x + h2*x^2 + h3*x^3 + ... + hn*x^n where * are...

Hi everyone... I'm a newbie... I have a question. I have to make something like y = h1*x + h2*x^2 + h3*x^3 + ... + hn*x^n where * are convolution. (volterra series - diagonal -) the signal x is in a file. Imagine I have the same file with 2 different bit per sample... 16bit e 32bit. When I make ^2 ^3 the values in 16 bit (-32768 +32767) increase the values in 32bit (-1 +1) decrease... ...