## Forums Search for: Aliasing

## Aliasing and Sampling Prog question

inI want to show aliasing and sampling graphically with spectra by using LabVIEW (or similar).Here is what I am doing I have a ficticious...

I want to show aliasing and sampling graphically with spectra by using LabVIEW (or similar).Here is what I am doing I have a ficticious sampling freq which is my 'analogue' signal - lets say I make it 500MHz. I then pass white noise through a narrowband filter centred at say 10.7MHz bandwidth 200kHz. I take the FFT and see the spectrum and its negative image - no problem. I add a sine w...

## automatic detection of aliasing?

inHi all, I need your help on the following difficulties: I am designing something like "automatic" detection of aliasing,...

Hi all, I need your help on the following difficulties: I am designing something like "automatic" detection of aliasing, programmatically. For a given signal discrete time signal x(n), possibly infinite support, we know its values are all positive, they are absolutely summable, so DTFT exists and we know that the signal die down for large n, and it can be roughly deemed as zero fo...

## wavetable synth & anti-aliasing

inhello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch...

hello, i'm trying to program a wavetable-based synth in c++. I have some waveforms and different envelopes for them and want to switch between settings while playing. I use a 4096 samples buffer. My question is, how do i now implement anti-aliasing? I thought of multiplying envelope & waveform, and then lowpass and the resample with linear interpolation at a lower sample rate. Is it right, tha...

## Nyquist and rectangular waveforms

inThis is prompted by the discussion of aliasing in the thread "Higher upsampling with minimum phase downsampling produces more aliasing" by...

This is prompted by the discussion of aliasing in the thread "Higher upsampling with minimum phase downsampling produces more aliasing" by 'jungledmnc'. My question is based on the observation that sampled rectangular waveforms can be reproduced exactly, using a trivial D/A converter (strobed latch), with no need for an anti-alias / anti-image filter. The only requirement is that the sam...

## Anti-Aliasing filters with sigma-delta A/Ds

inDo you still require an anti-aliasing filter with sigma-delta A/D's? I am guessing yes but very low order. A/D I am testing seems to have an...

Do you still require an anti-aliasing filter with sigma-delta A/D's? I am guessing yes but very low order. A/D I am testing seems to have an excellent roll-off without any filter at all. Hardy

## help regarding impulse invariance method !

inHi, i hav just studied the impulse invariance method & bilinear transform. In Impulse Invariance there is "many to one mapping" and hence...

Hi, i hav just studied the impulse invariance method & bilinear transform. In Impulse Invariance there is "many to one mapping" and hence aliasing takes place. while in Bilinear Transform there is "one to one mapping" and hence no aliasing. Actually what is the "one to one mapping" & "many to one mapping"? how it is related to aliasing?

## Beginner's question on undersampling and aliasing

inHi all, I am a newbie to DSP. I have a huge number of data samples that I want to perform FFT on. But due to the enormous amount of the data...

Hi all, I am a newbie to DSP. I have a huge number of data samples that I want to perform FFT on. But due to the enormous amount of the data samples, I'd like to use only 1M samples and perform FFT. I know that this is severe undersampling and that it does not satisfy the Nyquist rate. But all I want is to prevent aliasing so that I can get correct information about the low frequency content...

## Multiple ADCs and aliasing

inI have two 1MHz ADCs and would like to interleave them by a half of the sampling cycle in order to get a single ADC with effective sampling...

I have two 1MHz ADCs and would like to interleave them by a half of the sampling cycle in order to get a single ADC with effective sampling frequency of 2MHz (I want to oversample the signal by as much as possible). But the aliasing properties of the data stream will be as if the signal was sampled at 1MHz because the ADCs are independent blocks and don't influence each other, right? Bes...

## Average the averages

inI am making an application where several inputs has to be filtered. The filtering is a simple moving average filtering with a variable...

I am making an application where several inputs has to be filtered. The filtering is a simple moving average filtering with a variable filter length. The problem is what happens when I connect these moving average filters in cascade? I know that I will get some aliasing, but how big is the aliasing problem? The scenario is as follows: The fastest sampling rate of the system is 100 Hz. I connec...

## Anti Aliasing of Arbitrary Waveforms

inWhat are the common method(s) for preventing aliasing of arbitrary waveforms generated within a DSP application? I understand that waveforms...

What are the common method(s) for preventing aliasing of arbitrary waveforms generated within a DSP application? I understand that waveforms supplied as analog signals and presented to an ADC must first be lowpass filtered to remove harmonics above the Nyquist frequency, but what about generated waveforms? I know about BLIT and it won't be appropriate for completely arbitrary waveforms. I'...

## Problem with Filtering .... resulting in Aliasing

inI acquired the data at 1Khz which is too high for my analysis purposes as i had to see the frequency response down on low frequency range below...

I acquired the data at 1Khz which is too high for my analysis purposes as i had to see the frequency response down on low frequency range below 25Hz. I have tried using matlab filter topic but when i am using decimate function in matlab (below), i am seeing the aliasing/folding affect. i really want to do just low pass filter the data and downsample to 25hz and take psd/ pwelch to show the...

## Simulation of Aliasing

inI ran across a problem in a DSP book that got me to thinking (the best kind= of problem). The author suggests that we can illustrate aliasing...

I ran across a problem in a DSP book that got me to thinking (the best kind= of problem). The author suggests that we can illustrate aliasing by sampl= ing an audio signal at a high frequency and then replacing samples with zer= oes to get a lower effective sampling frequency. For example, if we sample= at fs1 =3D 32 kHz and then keep every M =3D 4th sample (replacing the othe= r samples ...

## Sound Card Question

inHow does a sound card set its anti-aliasing filters? After all, you can program a sound card to read at say 44.1kHz or 22,050Hz or half of...

How does a sound card set its anti-aliasing filters? After all, you can program a sound card to read at say 44.1kHz or 22,050Hz or half of that again so how do the ani-aliasing filters change? Switched cap filters are sampled filters so they would not be good and digital filters are no good either as we need analogue filters jus before sampling. Shytot

## How I can design real time dsp system

inHello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth...

Hello everyone, I want to design a real time DSP system.Band of interest extends from o to 4kHz.I will use 12 bit ADC and 3rd order butterworth lowpass filter. What I want to know is how I can estimate the minimum stopband attenuation for the anti-aliasing filter, minimum sampling frequency and The level of the aliasing error relative to signal level in the passband for the estimated Amin an...

## Sampling a signal corrupted by AWGN

inLet us assume that we have a bandlimited signal (with maximum frequency f_n) corrupted by additive white Gaussian noise. Before we can sample...

Let us assume that we have a bandlimited signal (with maximum frequency f_n) corrupted by additive white Gaussian noise. Before we can sample this signal, we pass the signal through an ideal anti-alias filter with cut-off frequency f_c > =f_n to avoid noise aliasing. The output of the anti-aliasing filter is fed into a matched filter matched to the symbol rate, (1/T)> =2f_n (i used f_n here because

## OT: Spacial aliasing (or so they claim)

inA beautiful demo of many oscillators with a common sub-period: http://tinyurl.com/3ee2eya Jerry -- Engineering is the art of making what you...

A beautiful demo of many oscillators with a common sub-period: http://tinyurl.com/3ee2eya Jerry -- Engineering is the art of making what you want from things you can get.

## book recommendation please...

inHi all, I'm looking for a good textbook that serves as an introduction to anti- aliasing filters, Nyquist frequency, sampling rates, SNR, low...

Hi all, I'm looking for a good textbook that serves as an introduction to anti- aliasing filters, Nyquist frequency, sampling rates, SNR, low pass filter design, etc. Can someone please suggest a reference? Thanks in advance, -weg

## depicting the concept of aliasing

inSorry for this supposedly simple question. In Richard Lyon's "Understanding Digital Signal Processing" p.27 is there a y axis for Figure 2-2(c)?...

Sorry for this supposedly simple question. In Richard Lyon's "Understanding Digital Signal Processing" p.27 is there a y axis for Figure 2-2(c)? And if so, what is the unit of that y-axis? Thank you.

## Sampling Question

inSuppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60...

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60 secs...so how is this possible? The values of the capacitors etc would be hugh. Of course I could over-sample1000s of times...is this the norm? K.

## 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 ...