Hi all, I was wondering how can I calculate data rate given: -B: bandwidth -cr: code rate (say convolution code) -some modulation scheme...

Hi all, I was wondering how can I calculate data rate given: -B: bandwidth -cr: code rate (say convolution code) -some modulation scheme assume the sinr is good enough thanks, Yiu

Hello, The scenario: A white noise signal. (Uniformly distributed between 0 and 1) A square pulse signal in the time domain. The first zero...

Hello, The scenario: A white noise signal. (Uniformly distributed between 0 and 1) A square pulse signal in the time domain. The first zero of the sinc function in the amplitude spectrum is at 50 Hz. The result of the convolution of the two above signals is a gaussian distributed signal. How can I predict that? Thank you!

I have images that I'm either trying to downsample or upsample, preferably using time-domain convolution with a FIR kernel. The...

I have images that I'm either trying to downsample or upsample, preferably using time-domain convolution with a FIR kernel. The main characteristic of these images is that they're overwhelmingly black, that means, isolated bright features spread around a very dark surface, quite similarly to the picture of a starry night sky. Here's the problem, due to the overwhelming presence of black are...

Revised after peer review, approached with a little less haste So... Sampling with a period of T is given by (after asciification) as...

Revised after peer review, approached with a little less haste So... Sampling with a period of T is given by (after asciification) as .. (1/T)sum (n : 0, inf)(d(t-nT) * f(nT) ) ... with * representing multiplication and not convolution as we are still in the time domain. However, (and this is where my protest came in having previously fully revised Fourier, Laplace, Butterworth...

I managed to get hold of a full copy of Signal Wizard 2 which computes the FIR coefficients of arbitrary filters. I'm designing some HP filters...

I managed to get hold of a full copy of Signal Wizard 2 which computes the FIR coefficients of arbitrary filters. I'm designing some HP filters with 20 degree phase lags for surround sound work and want to use the coefficients exported by the program in my own routines. I've got a load of MP3 files that I want to process, and don't want to use MATLAB 'cause it has to be a small stand-alone pa...

What's a kernel? Where a dog lives! I hear many people using the word here on an engineering group. It never got mentioned when I was a student...

What's a kernel? Where a dog lives! I hear many people using the word here on an engineering group. It never got mentioned when I was a student of engineering even though we did a fair amount of maths. My friends who were maths people knew all about it though. It it the same as impulse response in a discrete convolution or is its generalisation to time-varying systems?Is there an engineering...

I have a vibroseis dataset which has been collected in the field. To my knowledge, the recorded signal s'[t] is the convolution of the...

I have a vibroseis dataset which has been collected in the field. To my knowledge, the recorded signal s'[t] is the convolution of the original sweep s[t] and the reflection response r[t] of the layered earth. Thus, the relationship is expressed in its simplest form as s'[t] = s[t]*r[t], neglecting the effects of earth attenuation and noise. However, I do not have the reference sweep s[t] since...

Hello, Signal Processing Experts, There is something that I have been so used to and does not look so obvious today. On page 98 of...

Hello, Signal Processing Experts, There is something that I have been so used to and does not look so obvious today. On page 98 of Communication Systems Engineering by Proakis (1st edition), it defines the autocorrelation function as the convolution between the signal itself and time reversed version of its complex conjugate: Rx(tau) = x(tau) convolve x*(-tau) ...

Hi, i am new to this newsgroup and relatively new to dsp programming (i have basic theoretical background). For my application im trying to...

Hi, i am new to this newsgroup and relatively new to dsp programming (i have basic theoretical background). For my application im trying to implement i need to realtime convolute an audiostream with a time-varying filter. Is there some opensource or at least free C/C++ library that i can easyly use for that? (im alreay checking http://www.fftw.org/ for my fft needs, but would like a lib ...

I'm looking forward implementing image interpolation in one dimension at once, however I don't really know how to do it. Linear interpolation...

I'm looking forward implementing image interpolation in one dimension at once, however I don't really know how to do it. Linear interpolation doesn't fit me good, I need something better, more like cubic spline interpolation, but the only problem that well implementing it doesn't sound like such an easy thing (I will if I have to, but it definitely won't be my first choice for that precise ...

I've got myself befuddled here..... Start with a temporal cosine of peak amplitude 1.0 with an integral number of cycles in 256 seconds-...

I've got myself befuddled here..... Start with a temporal cosine of peak amplitude 1.0 with an integral number of cycles in 256 seconds- period of some integer multiple "k" of 1/256Hz like 10/256. Window it (multiply it) with a rectangular window of length 256 and amplitude 1.0 - from t=0 to t=255. This does nothing to the original samples within that time span. So, we have multiplied in ...

Instead of just taking the convolution of a signal with the reference or kernel once and then taking the square root in the frequency domain,...

Instead of just taking the convolution of a signal with the reference or kernel once and then taking the square root in the frequency domain, why not match filter with the reference again and again to recover the original clean signal? Well, you _can_ do that if a wave form is all you want, but you can get that from the ref. alone _anyway_. If you want to keep the original magnitude of t...

I have the absolute amplitude of a digital signal that I want to auto correlate with self , also cross correlate with a different digital...

I have the absolute amplitude of a digital signal that I want to auto correlate with self , also cross correlate with a different digital signal [ absolute magnitude ] . I'm using a complex 1d fft to implement this . I only have the absolute values of the spectra to work with ; where the absolute value is defined as : square root of (Real^2 + Imaginary^2). After multiplying the a...

Hi, I am trying to design an OFDM trasmitter. My current structure is as follows: IFFT -> Cyclic Prefix insertion on I and Q -> x4...

Hi, I am trying to design an OFDM trasmitter. My current structure is as follows: IFFT -> Cyclic Prefix insertion on I and Q -> x4 interpolation filter on I and Q -> I/Q quadrature modulator -> output to DAC. My question is related to the interpolation filters. The convolution between the input to the filter and the filter impulse response smudges the symbol. For example, if the FFT

hi please help me on this x(t)=sin(2*pi*10*t); w(t)--> awgn noise y(t)=x(t)*e(j*w(t)); here x is multiplied with e(j*w) in time domain. that...

hi please help me on this x(t)=sin(2*pi*10*t); w(t)--> awgn noise y(t)=x(t)*e(j*w(t)); here x is multiplied with e(j*w) in time domain. that is equivalent to convolution if frequency domain how to estimate e(j*w(t)) using deconvolution in frequency domain?

Hai, I am using a 1/2 rate convolution encoder and after encoding I am mapping the bits to qpsk constellation.My received signal is of form...

Hai, I am using a 1/2 rate convolution encoder and after encoding I am mapping the bits to qpsk constellation.My received signal is of form y = a*s+n. Where 'a' is due to channel fading 's' is qpsk symbol and 'n' is noise.I estimated 'a' (I know 'a') .I have no idea about soft qpsk demapper. that will take N symbols and outputs 2*N coded soft bits. Thank You M.Rakesh

Hello, I am simulating the performance of an OFDM based communication system, where I convolve OFDM symbols with the Channel Impulse Response...

Hello, I am simulating the performance of an OFDM based communication system, where I convolve OFDM symbols with the Channel Impulse Response (7-Tap Rayleigh distributed) The problem is that Matlab function conv( ) takes a lot of processing time. I tried to declare the output variable as a vector of zeros. This did help, but only slightly. Under Monte Carlo's method, it gets really annoyi...

Here's my problem. I have an image to be processed (well, deconvolved pretty much), and a star-shaped impulse response I want to deconvolve it...

Here's my problem. I have an image to be processed (well, deconvolved pretty much), and a star-shaped impulse response I want to deconvolve it with. The impulse response is symmetric in both the vertical and horizontal axis and pretty much centred, but not very precisely. Of course both the image and the impulse response are appropriately zero- padded as to avoid any circular convolution. ...

Hi pals, My post title is maybe very weird, but let me expose the situation: I want to filter a real infinite signal x(k) by block...

Hi pals, My post title is maybe very weird, but let me expose the situation: I want to filter a real infinite signal x(k) by block convolution. To do this I have the the frequency response of an IIR filter (called h). The frequency response correspond to : H(k) = B(k)./A(k) (matlab notation ./ => pointwise division) where B(jk) is the 1024pts DFT of a sequence b(k) which has 128pts, same fo

So you model a square wave by the sum of sine waves. Suppose you take 5 or 6 and get the classic Gibbs phenomena. Now suppose that you know...

So you model a square wave by the sum of sine waves. Suppose you take 5 or 6 and get the classic Gibbs phenomena. Now suppose that you know exactly where the 5th or 6th harmonic is and take the same square wave and pass it through a second - or higher order filter (analogue). You don't get Gibbs phenomena at all. All you get is the transient response (ie the convolution of the squar