## Spatial Aliasing in Beamformers

Started by in comp.dsp15 years ago 6 replies

I have been reasing a paper that says that for acoustic beamformers (I suppose the same applies to EM beamformers), the distance...

I have been reasing a paper that says that for acoustic beamformers (I suppose the same applies to EM beamformers), the distance between microphones must be d

## Aliasing - A new perspective out of Box

Started by in comp.dsp16 years ago 33 replies

HI all, Thank you very much for the participants in " Negative Frequencies" Thread. Here is another question from me. Yesterday I was...

HI all, Thank you very much for the participants in " Negative Frequencies" Thread. Here is another question from me. Yesterday I was watching a movie. In that, hero was chasing villain in a BMW car at a speed of 100 Miles/Hr.I was observing the car keenly and suddenly it appeared that the tyres of the car are rotating at only 8 miles/hr(I didnt measure..felt that its RPM is very less)...

## sampling ...

Started by in comp.dsp6 years ago 6 replies

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a...

Sirs, I have some questions related to sampling and subsequent re-construction of signals. I am hsitoping to get answers ... 1) Assume a signal consists of two signals of frequencies f1 and f2 such that frequncy f1 < f2. Sampling is done to ensure fs > = 2xf1 and fs < 2xf2. Since the sampling rate does not meet the nyquist rate for f2, it would lead to f2 aliasing into f1. When the discre

Started by in comp.dsp15 years ago 5 replies

Thanks for not losing your patience. I am going to make a resume of what my intentions are and list my two questions in order of priority. I...

Thanks for not losing your patience. I am going to make a resume of what my intentions are and list my two questions in order of priority. I receive your postings with a latency of 5 hours so if someone has already made postings giving me solutions please discard these message. Question 1: I need to compute the aggregate frequency response of three cascaded multirate filters. I was tryi...

## Multirate filters

Started by in comp.dsp11 years ago 1 reply

hi all, I am working on polyphase filter bank design. I want some pointers on (i)how to avoid aliasing from other bands? (ii)what are the...

hi all, I am working on polyphase filter bank design. I want some pointers on (i)how to avoid aliasing from other bands? (ii)what are the important steps while considering the design of prototype filter? (iii)How to avoid the amplitude distortion? (iv)How to design filter banks with different passband widths? thanks in advance vanamali

## Approaching QAM256 demodulation

Started by in comp.dsp8 years ago 14 replies

Hi! I am trying to write a demodulator for a 6MHZ-wide QAM256 signal. I have captured a sample file with the aid of a USRP, and I carry on...

Hi! I am trying to write a demodulator for a 6MHZ-wide QAM256 signal. I have captured a sample file with the aid of a USRP, and I carry on the rest in Matlab. [Question regarding sampling frequency]: I get max 8MS/s from the USRP (although the spec says it should support 16MS/s - I can't seem to get that for some reason). It seems like I should take > 12MS/s to avoid aliasing for a 6MHz si

## Frequency Domain represeatation

Started by in comp.dsp14 years ago 7 replies

Guys: Why do the charts of the textbook representing the frequency response of a filter always show the range from -pi to pi instead of from 0...

Guys: Why do the charts of the textbook representing the frequency response of a filter always show the range from -pi to pi instead of from 0 to 2pi? If use the range from 0 to 2pi, examples in the chapter 2 of Understanding Digital Processing by Lyons would cause big problem. The sampling replication would shift to the right and aliasing explaination failed. Why "-pi to pi" not "0 to 2pi...

## Zero padding and Cross Correlations

Started by in comp.dsp14 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...

## Phase in FIR

Started by in comp.dsp12 years ago 26 replies

Hi, I'm trying to specify a DSP system to measure a two sine waves around 100kHz and their resulting harmonics down to the 5th. The phase...

Hi, I'm trying to specify a DSP system to measure a two sine waves around 100kHz and their resulting harmonics down to the 5th. The phase information is important as well as the magnitude of the harmonics. In order to avoaid any aliasing and phase problems with an analog filter I 'm intending to oversample at 4MSPS. From what I can tell I will then need to perform a low pass FIR bef...

## Filter Bank theory

Started by in comp.dsp10 years ago 3 replies

Hello, I am reading a paper about filter bank theory and I need help in understanding something. Say a filter bank (FB) contains M analysis...

Hello, I am reading a paper about filter bank theory and I need help in understanding something. Say a filter bank (FB) contains M analysis and M synthesis filters. Since the effective bandwidth of each subband signal is pi/M, it can be downsampled to reduce the total rate. How can downsampling not introduce aliasing? For example if Fs = 360 and one of the bands is 240-270Hz, then wou...

## Continuity in frequency domain, post processing to avoid time-domain aliasing

Started by in comp.dsp6 years ago 4 replies

Hi list, After a while without dealing with FFTs (audio processing), I have some ideas that I would like to implementÂ… but first of all I'd like...

Hi list, After a while without dealing with FFTs (audio processing), I have some ideas that I would like to implementÂ… but first of all I'd like to clarify some of the doubts that I have never really solved. I write some stuff to arrive to the point, please correct me if I am wrong. Everytime I do some FFT analysis and resynthesis using constant overlap and add I end up with hearin

## Power distribution across aliased signals

Started by in comp.dsp11 years ago 8 replies

I am having trouble understanding the effect of aliasing/imaging on the power of an output signal. Imagine I am using a DAC to generate a single...

I am having trouble understanding the effect of aliasing/imaging on the power of an output signal. Imagine I am using a DAC to generate a single frequency sine wave with frequency f well below fs/2 and with power P. Assume there is no reconstruction filter. So this frequency will image repeatedly across the frequency domain at n.fs-f and n.fs+f. I do not understand how the original power, P, i...

## decimation and anti-aliasing

Started by in comp.dsp12 years ago 4 replies

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture...

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture of what I am trying to do. I have a large number (order of 10^13) of discretized data samples. I have to obtain the frequency spectrum of this data so that I can predict minima in the discretized data. I am using the FFTW C library to obtain the Frequency...

## Why is it bad to have spectral samples nonzero at fs/2?

Started by in comp.dsp14 years ago 21 replies

I'm working on a paper about interpolation that I threatened to do long ago. I'm trying to say that a spectrum that has nonzero (or not small)...

I'm working on a paper about interpolation that I threatened to do long ago. I'm trying to say that a spectrum that has nonzero (or not small) samples at or near fs/2 is problematic. But I'm having a bit of trouble saying why it's bad necessarily. One thing one can say is that it's likely that the spectrum was not properly bandlimited before sampling and spectral aliasing is likely to h...

## Re: Interpolation

Started by in comp.dsp11 years ago 9 replies

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the...

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the sinc is scaled wider in time for > lowpass responses narrower than the Nyquist band. that's a choice that someone might make. > This would be done > for any lowpass anti-aliasing filter designed for sample rate > reduction with this me

Started by in comp.dsp11 years ago 16 replies

Hi, I understand that zero-padding a time domain signal cannot add any extra information to the spectrum, and that you only end up with a more...

Hi, I understand that zero-padding a time domain signal cannot add any extra information to the spectrum, and that you only end up with a more finely sampled spectrum. What I'm interested to know is whether zero-padding can cause spectral leakage? Or if I will always see a true representation of my original signal in the spectrum? Basically, I have been zero-padding my data by 100,000 poi...

## Absolute Beginner - inverted signal?

Started by in comp.dsp11 years ago 24 replies

Hi Guys, First please excuse my ignorance, I am just starting in DSP as a bit of a hobby. I Have Richard Lyons "understanding dsp" here and...

Hi Guys, First please excuse my ignorance, I am just starting in DSP as a bit of a hobby. I Have Richard Lyons "understanding dsp" here and am working my way through the "Sampling Bandpass Signals" chapter. I understand the idea of Aliases and aliasing in the sampled signal spectrum but in the diagrams for the continuous signal spectrum there is always an inverse of the signal. So say there ...

Started by in comp.dsp13 years ago 1 reply

Hi, I am new to the whole DSP thing (very new) and have a question regarding the use of undersampling and quadrature mixing to demodulate a...

Hi, I am new to the whole DSP thing (very new) and have a question regarding the use of undersampling and quadrature mixing to demodulate a signal in a given range. I have been trying to research the topic but have not been getting very far. I understand how the bandpass signal can be undersampled at a rate of 2B without aliasing if the proper bandedge frequencies or chosen but have no real ...

## Re: window function-dsp

Started by in comp.dsp8 years ago 1 reply

On Wednesday, January 19, 2011 2:25:07 AM UTC-5, rashmi venugopal wrote: > > Bharat: > > Total harmonic distortion can be measured in many...

On Wednesday, January 19, 2011 2:25:07 AM UTC-5, rashmi venugopal wrote: > > Bharat: > > Total harmonic distortion can be measured in many ways, but Rashmi asked > ab= > > out filter design. (A filter to suppress aliasing is needed even for your > F= > > FT approach.) > > > > Rashmi: > > Windowed-sinc design is historically interesting, but rarely the best > desig= > > n method when a

## Aliasing of Quad-detector sensor signals

Started by in comp.dsp5 months ago 8 replies

I realise DSP engineers don't celebrate Xmas so I feel warranted in asking a question at this time of year. A quad-detector connected to say the...

I realise DSP engineers don't celebrate Xmas so I feel warranted in asking a question at this time of year. A quad-detector connected to say the shaft of a motor gives out pulses and senses direction. So it effectively gives out + or -1 which we count in the main loop which must execute at say within Ts seconds. Suppose there are thousands of pulses per revolution and my understand