## Forums Search for: Aliasing

## OT:Sampling in Stats

inWhen the Stats people sample data for analysis they have a load of rules for population size etc. However, they never have to filter the data...

When the Stats people sample data for analysis they have a load of rules for population size etc. However, they never have to filter the data first to avoid aliasing. Is this because the data is already in "digital" format? Hardy

## PCM 3003

inDear All Anybody knows how to change the sampling rate of PCM3003(Daughter Card for C6711 DSK card), it is 48Ksps and I need to work with...

Dear All Anybody knows how to change the sampling rate of PCM3003(Daughter Card for C6711 DSK card), it is 48Ksps and I need to work with 8ksps, apparently it has internally anti aliasing and post processing filter for 48Ksps. Thanks: Hamid

## Analyzing an "undersampled" sequence

inPerhaps I should post this elsewhere but we speak the same language here. I may have asked a similar question some time ago but now I have a...

Perhaps I should post this elsewhere but we speak the same language here. I may have asked a similar question some time ago but now I have a new perspective and want to investigate. I have a wastewater process that's being sampled periodically (uniform sampling for what it's worth). The sample rate is way too low to avoid aliasing but the samples are real enough and the data is contin...

## zero padding avoids aliasing

inhey guys - i can't seem to find find this statement any where - but it makes sense to me - could someone please verify. When you find the DFT of...

hey guys - i can't seem to find find this statement any where - but it makes sense to me - could someone please verify. When you find the DFT of a signal say: x[0]=2, x[1]=4, x[2]=6, x[3]=8 - you will get four discrete points in frequency (per period) now if you zero pad such that tou get x[0]=2, x[1]=4, x[2]=6, x[3]=8 x[4]=0, x[5]=0, x[6]=0, x[7]=0 - you will once agin get four discrete frequenc...

## Got it to work :-)

Hello all. I'm very happy to say my filter is working perfectly. What it looked like before was that I was aliasing input frequencies above...

Hello all. I'm very happy to say my filter is working perfectly. What it looked like before was that I was aliasing input frequencies above Fs/4 the decimating Nyquist. After much trial and error I think my problem was that 1. I didn't account for the filter gain - after filtering - the skirts then were heavily amplified. 2. I may have not added in correctly the previous MAC. (time for vers...

## bypassing antialiasing filter of TLV320AIC23

inHello, I have been using C6713 DSK. My question is; Is there antialiasing filter in front of AIC23 codec of DSK. If yes, can bypass...

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.

## Best FIR Coefficients for multirate app

inHello for this newbie question! I'm on developing a simple audio-sequencer application. So, if I want to play a 44100Hz samplesound with a...

Hello for this newbie question! I'm on developing a simple audio-sequencer application. So, if I want to play a 44100Hz samplesound with a default note at D4 like a drum at lower notes like C4,F3,E3,etc., I have to decimate to the according sampling-rate. But of course, to prevent aliasing, there must be a lowpass filter before decimation. The best way to to this very efficient is an FIR-...

## producing odd&even harmonics

inHi, This is a little bit a repost but i have some new information now. I'd like to produce harmonics in a signal. Using the chebychev...

Hi, This is a little bit a repost but i have some new information now. I'd like to produce harmonics in a signal. Using the chebychev polynomials, you can produce individual overtones (1,2,3th, 4th, ... harmonic ) where T1(x) = 1; T2(x) = x T_n(x) = 2xT_{n-1} - T_{n-2}. But how to produce the undertones? Can i base myself on aliasing for this effect? how pls? Help ap...

## Newbie question

inI have a 2d signal (an image, actually) of which I want to calculate ACF. I tried to use the IFFT/FFT approach, but when I rotate the image...

I have a 2d signal (an image, actually) of which I want to calculate ACF. I tried to use the IFFT/FFT approach, but when I rotate the image I notice some strange effects (something aliasing-like). So I tried to pad my image both vertically and horizontally with zeros, doubling its dimension before calculating the FFT. Am I doing right? Someone can help me? Thank you Lorenzo

## antialiasing and decimation

inHi, is there a way to decimate a signal (reduce sampling rate) reducing aliasing without implementing a computationally intensive...

Hi, is there a way to decimate a signal (reduce sampling rate) reducing aliasing without implementing a computationally intensive low-pass filter? Consider that this signal is supposed to have frequency components higher than the reduced sampling rate. thanks

## Question about aliasing formula in Lyons DSP

inHi Everyone. I am a self taught programmer who also designs hardware, and I am trying to get up to speed on DSP with Lyon's "Understanding...

Hi Everyone. I am a self taught programmer who also designs hardware, and I am trying to get up to speed on DSP with Lyon's "Understanding DSP". I've posted a snapshot from the book here: http://i.imgur.com/MSGVtEY.jpg I am following everything up through equation 2-3. "m" is any integer, so adding 2PIm gives the same sin value. What I don't get is how he then restricts the meaning of ...

## QMF filter Order

Hi, I have read in many books that the order of the filters used in QMFs should be odd to satisfy the aliasing canceling condition. Can...

Hi, I have read in many books that the order of the filters used in QMFs should be odd to satisfy the aliasing canceling condition. Can anyone explain how the filter order is related to the above mentioned condition. Thanks

## Pitch shifting question

inHi. I want to do some pitch shifting of some samples I have. I'm trying to avoid the whole sample rate conversion problem. I can hack up...

Hi. I want to do some pitch shifting of some samples I have. I'm trying to avoid the whole sample rate conversion problem. I can hack up the hardware some. Q: Can I just alter the clock and rate that I send data to my DAC *WITHOUT* changing the anti-aliasing low-pass and accomplish my goal? Thank you. Henry.

## Basic Questions??

inHello 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...

## Decimating

inHello 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...

## Acoustic Beamforming Question

inI always assumed that sensors for a beamformer had to be spaced at least lambda/2 (half a wavelength) apart else we get spatial aliasing. Is...

I always assumed that sensors for a beamformer had to be spaced at least lambda/2 (half a wavelength) apart else we get spatial aliasing. Is this true for acoustic arrays too? I have seen papers on small arrays (circular say) for mobiles and such like. Let us asume that the mics are 1cm apart giving a wavelenth of 0.02m and a frequency (min) of 16.5kHz. Not much use is it? Hardy

## FFT: Could you help me to clarify some doubts?

inHello I should calculate a FFT of signal that have max 500Hz of frequency. I know that for avoid aliasing problem I must have a sample rate of...

Hello I should calculate a FFT of signal that have max 500Hz of frequency. I know that for avoid aliasing problem I must have a sample rate of Fmax*2, so I have fix it to: Sample_Rate_Frequency= 1024Hz Question 1) Is correct the follow table: [Acquisited Point] [Acquisition Duration] [Frequency Resolution] 1024 1 sec 1H...

## sampling problem

inDear 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 ...

## 10MHz signal sampling - which DSP?

inHello, i would like to process an analog signal, coming out from a photodiode with a maximum frequency content of 7-10MHz. Here my idea...i...

Hello, i would like to process an analog signal, coming out from a photodiode with a maximum frequency content of 7-10MHz. Here my idea...i don't know if it is something correct or not... i need your help... 1. A/D conversion, trying to aversampling the signal (at least 40-50MSPS) - anti aliasing filter to be evaluated (9-10 bits are enough) 2. CPLD of small FPGA after the DAC (maybe a Ma...

## Upsampling problem

inHi, 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...