Basic Questions??

Started by Jamie in comp.dsp14 years ago 23 replies

Hello to all I am an Engineering student and in my Viva Exam we were asked some questions which are as follows:: (1). How to acquire or...

Hello to all I am an Engineering student and in my Viva Exam we were asked some questions which are as follows:: (1). How to acquire or sample an arbitrary signal such that Aliasing is completely avoided ??We dont know anything about the Input. (2). How can we generate Impulse Waveform from our computer so that we can use those impulse waveform in our DSP Hardware for doping an Impuls...


Re: bypassing antialiasing filter of TLV320AIC23

Started by Rune Allnor in comp.dsp13 years ago

On 15 Jun, 17:37, murselonder wrote: > Hello, > > I have been using C6713 DSK. My question is; Is there antialiasing > filter in front...

On 15 Jun, 17:37, murselonder wrote: > Hello, > > I have been using C6713 DSK. My question is; Is there antialiasing > filter in front of AIC23 codec of DSK. If yes, can bypass (omit, > cancel) it? > > In other words, can I sample over 48 kHz frequencies with aliasing > intentionally. You might want to ask this question on comp.dsp. I have cross-posted this rep


Decimating

Started by Luis Fernando in comp.dsp16 years ago 10 replies

Hello I'm trying to implement a distortion effect (following http://music.calarts.edu/%7Eglmrboy/musicdsp/FAQs/guitar_distortion_FAQ.html )...

Hello I'm trying to implement a distortion effect (following http://music.calarts.edu/%7Eglmrboy/musicdsp/FAQs/guitar_distortion_FAQ.html ) with 48000 Hz sampling and minimum "useful" frequency = 20 Hz. My problem is with aliasing. To minimize its effect, I want to oversample, apply the distortion and then decimate it. The idea seens simple and straighforward, but what's the buffer size...


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

Started by ale.saccoia in comp.dsp7 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


sampling problem

Started by walala in comp.dsp16 years ago 8 replies

Dear all, When I learned sampling theory I did not have any problem with it. But recently some questions always haunted among my minds... I...

Dear all, When I learned sampling theory I did not have any problem with it. But recently some questions always haunted among my minds... I need your help... I guess everything we sense is what have been sampled. I become suspect our sensing organs. Our vision seems to have no problem, very clear in seeing objects, but is it possible that our eyes only sampled an aliasing version of the ...


Upsampling problem

Started by jungledmnc in comp.dsp12 years ago 25 replies

Hi, I'm performing upsampling for audio to avoid aliasing. This is what I do: 1) Take source buffer and using cubic interpolation convert it...

Hi, I'm performing upsampling for audio to avoid aliasing. This is what I do: 1) Take source buffer and using cubic interpolation convert it into a buffer let's say 16x larger. 2) Perform lowpass on the temporary buffer with cutoff at X / 16, where X is just some kind of factor compensating steepness of the filter. 3) Here comes the blackbox - some effect. But in this testing case it simply do...


Purpose of interpolation in DACs?

Started by Ben Jackson in comp.dsp14 years ago 2 replies

I'd like to test my understanding of the usefulness of interpolation in a DAC: If you feed a signal to a DAC at Fc < Fclk/2 then there is a...

I'd like to test my understanding of the usefulness of interpolation in a DAC: If you feed a signal to a DAC at Fc < Fclk/2 then there is a primary image at Fc and (possibly unwanted) aliasing at Fclk +/-Fc and at harmoics of Fclk +/-Fc. Now, if you upsample (zero-stuff) the input to a higher Fclk, then the DAC runs faster, and the DAC-produced images are now at n*Fclk+/-Fc. However, th


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

Started by Fred Marshall in comp.dsp16 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...


Approaching QAM256 demodulation

Started by haxorcize in comp.dsp9 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


Beating Nyquist?

Started by Andor in comp.dsp13 years ago 48 replies

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this...

Friends, just now I stumbled upon this webpage: http://www.edi.lv/dasp-web/ Specifically, in this chapter http://www.edi.lv/dasp-web/sec-5.htm they state that they can sample a 1.2GHz signal using a pseudo-random sampling instants with an average rate of 80MHz (in the last line of section "5.2 Aliasing, how to avoid it"). I know that for nonuniform sampling, a generalization of...


Absolute Beginner - inverted signal?

Started by BobTheDog in comp.dsp12 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 ...


Creating brick-wall anti-aliasing filters?

Started by Funky in comp.dsp16 years ago 3 replies

How about using a Cauer with it's poor phase reponse,but the software compensating for it? Consider the following:- 1. An analogue...

How about using a Cauer with it's poor phase reponse,but the software compensating for it? Consider the following:- 1. An analogue filter is cascaded with an N-bit ADC feeding an N bit DAC. 2. An analogue filter is cascaded with an N-bit digital filter designed to create a response equal to that above. Will there be a difference between the two responses? Suppose that for the abo...


Spatial Aliasing in Beamformers

Started by Tom in comp.dsp16 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


Multirate filters

Started by maverick.gvs in comp.dsp12 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


Inverse FIR computation

Started by Shafik in comp.dsp16 years ago 6 replies

Hello all, I am trying to compute the inverse of an FIR filter: something that would reverse the effect of the first filter. I just want to...

Hello all, I am trying to compute the inverse of an FIR filter: something that would reverse the effect of the first filter. I just want to make sure my steps are correct: - take the FFT of the FIR - compute complex 1/f (in the frequency domain) - invert back into the time domain. (IFFT) - circular shift to adjust for the aliasing - window the data Does that seem right? --Shafik ...


Help! A puzzlement about noise sampling & reconstruction.

Started by Qian...@gmail.com in comp.dsp12 years ago 4 replies

a bandlimit white noise x(t) with PSD of S0 is sampled (no aliasing) to produce x[n]. The PSD of x[n] is calculated to be S0/Ts (Ts is...

a bandlimit white noise x(t) with PSD of S0 is sampled (no aliasing) to produce x[n]. The PSD of x[n] is calculated to be S0/Ts (Ts is the sample period). Now I just reconstruct the continuous noise xr(t) by passing x[n] impulses to the ideal reconstruction filter (gain=Ts, -fs


sampling ...

Started by manishp in comp.dsp7 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


Cascaded Integrator Comb and aliasing experts please help!

Started by David Joseph Bonnici in comp.dsp16 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...


Zero padding and Cross Correlations

Started by dspchick in comp.dsp15 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...


Filter Bank theory

Started by ayyaz ayyaz in comp.dsp11 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...