## Forums Search for: Aliasing

## Aliasing of Quad-detector sensor signals

inI 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

## aliasing condition for radon transform?

inI am talking about geophysical data here, where you have a record consisting of a few hundred traces, so the data has two dimensions time and...

I am talking about geophysical data here, where you have a record consisting of a few hundred traces, so the data has two dimensions time and distance (or t-x in short). If an event has a dip or slowness less than one sample per trace, it is not aliased. If it has greater dip, it would alias, unless the temporal frequency content is limited. Now radon transform can be used to remove certa...

## Re: bypassing antialiasing filter of TLV320AIC23

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

## Is this 'clever' method of filtering legit?

inI'm getting strange results from 'clever' filtering -- is what I'm doing legit? There's often a moire-like pattern in the output, even though...

I'm getting strange results from 'clever' filtering -- is what I'm doing legit? There's often a moire-like pattern in the output, even though the filtering tamps down the aliasing as intended. My source is digitized acoustic data replayed from wave files. The replay rate can be 1x, 2x, 4x or 8x. For rates > 1x I filter the data (and desample) to knock down frequencies that would alias into t

## Re: Questions about equivalents of audio/video and digital/analog.

inOn Sun, 19 Aug 2007 23:26:16 -0700, dplatt@radagast.org (Dave Platt) wrote: > "Digital" and "subject to aliasing" are two different things. >...

On Sun, 19 Aug 2007 23:26:16 -0700, dplatt@radagast.org (Dave Platt) wrote: > "Digital" and "subject to aliasing" are two different things. > > As I believe the term "digital" is usually meant, it implies a > two-state (on/off) storage representation. It's not just that the > signal amplitude is quantized, but that the quantization uses a > power-of-two representation and storage system of so

## Re: AM digital demodulation using the absolute value

I don't think you are grasping the idea of the low pass filter. It is to r= eject the carrier, not just the aliasing. You want the modulated...

I don't think you are grasping the idea of the low pass filter. It is to r= eject the carrier, not just the aliasing. You want the modulated signal wi= thout the carrier. So the cutoff frequency of the filter would be above yo= ur modulating signal frequency and below the carrier frequency. Everything= else you decide will depend on the details of your problem. =20 BTW, you can do bett...

## Purpose of interpolation in DACs?

inI'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

## Re: window function-dsp

inBharat: Total harmonic distortion can be measured in many ways, but Rashmi asked about filter design. (A filter to suppress aliasing is needed...

Bharat: Total harmonic distortion can be measured in many ways, but Rashmi asked about filter design. (A filter to suppress aliasing is needed even for your FFT approach.) Rashmi: Windowed-sinc design is historically interesting, but rarely the best design method when appropriate software is available. Windows suppress unwanted responses in the stopband, but they increase the leng

## wide band aliasing noise in the spectrum of non-uniformly sampled signal

inHello, As you know that whenever we perform GDFT( General Discreat Fourier Transform), on a nonuniformly sampled data, we obtain...

Hello, As you know that whenever we perform GDFT( General Discreat Fourier Transform), on a nonuniformly sampled data, we obtain wideband alysing noise alongwith signal spectrum, which donot allow us to analyse the signal with apprepriate approximation. If any one have some idea how to solve this problem, please tell me. I shall be thankfull to you for your help. Qaisar ...

## Interchanging of filtering and decimation operations

inHi, It is known that downsampling introduces aliasing if the signal to be downsampled has high frequency components. Hence it is passed through...

Hi, It is known that downsampling introduces aliasing if the signal to be downsampled has high frequency components. Hence it is passed through a low pass filter before it is downsampled. I want to know whether the filtering and the downsampling operations can be interchanged as they are essentially the same, or is it always necessary to filter the signal first and then downsample it. Thanks i...

## Inverse FIR computation

inHello 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 ...

## Anti-Aliasing filter

inOk I am sampling with an ordinary A/D (not sigma-delta) at 11025kHz which mean I need any noise to be attenuated at half sampling which is at...

Ok I am sampling with an ordinary A/D (not sigma-delta) at 11025kHz which mean I need any noise to be attenuated at half sampling which is at 5512.5Hz. I have read that the nosie level needs to be less that the R.M.S Quantisation Level of my A/D. Now I swing +or - 10 volts with 16 bits. So my Quantisation level is delta = dynamic range/2^16=20/65536=0.000305176 volts. (or 0.305mV) Now...

## Anti-aliasing advice for de-interleaved signals

inHi everyone - this is my first post here. It's about time I joined a DSP forum. I need some advice related to a measurement system. The system...

Hi everyone - this is my first post here. It's about time I joined a DSP forum. I need some advice related to a measurement system. The system samples at 1 kHz but between every sample the input to the DAQ is switched between two inputs. The result is that the acquired signal, at 1 kHz, is two interleaved signals at 500 Hz each. I would then like to go on to separate the two signals, which a...

## help with complex decimation and band shifting

inHi there, I am working in the frequency domain and this must be done in the frequency domain. I have a signal that I would like to decimate...

Hi there, I am working in the frequency domain and this must be done in the frequency domain. I have a signal that I would like to decimate by a factor of 4. Do I simply keep every 4th bin and keep it at that? Would I have to apply a lowpass filter in order to prevent aliasing? If so, how to do I determine the correct lowpass filter to use? After my complex decimation, I need to do a b...

## Creating brick-wall anti-aliasing filters?

inHow 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...

## Design of anti-alias filter for ADC, with oversampling and averaging

inHi, I am trying to design proper anti-aliasing filters for a new hobby project (digital control of a quad-rotor flying robot - see...

Hi, I am trying to design proper anti-aliasing filters for a new hobby project (digital control of a quad-rotor flying robot - see previous post for more questions). Here are the specifics: * My sensors are analog MEMS accelerometers and gyroscopes. I'll have at least 6 sensors to process, at least initially. * I will be using a microcontroller with 10-12 bit built-in ADCs to process t...

## Help! A puzzlement about noise sampling & reconstruction.

ina 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

## dealing with aliasing

inHello Forum, if a continuous-time sinusoid x(t)=cos(2pi*f1*t+theta) is sampled at an arbitrary rate f_s, we will obtain a sequence...

Hello Forum, if a continuous-time sinusoid x(t)=cos(2pi*f1*t+theta) is sampled at an arbitrary rate f_s, we will obtain a sequence x[n]. This sequence will be the same sequence we would obtain from sampling, at the same rate f_s, an infinite number of other sinusoids of continuous frequency different from f1. The spectrum would be a series of delta at locations f =f1 +- k*f_s. This would...

## Common practice - Aliasing in transition region?

inI'm designing a low pass FIR and want to minimise the number of taps - no surprises there. The FIR is an antialiasing filter in a DDC The...

I'm designing a low pass FIR and want to minimise the number of taps - no surprises there. The FIR is an antialiasing filter in a DDC The target bandwidth is 80% of the available bandwidth eg Fs = 125MHz Passband is 50MHz Nyquist = 62.5MHz When designing the filter is it reasonable to have the stop band start at 75MHz? My rationale for this is that any signals in the range 62.5 to 75MHz ...

## Nyquist Condition question.

inHi all, (1) I know the famous Nyquist Condition, f(t) with bandwidth B is sampled without aliasing if Fs> 2B. Today I read another...

Hi all, (1) I know the famous Nyquist Condition, f(t) with bandwidth B is sampled without aliasing if Fs> 2B. Today I read another thing called Nyquist Condition which says: for a continuous time signal x(t), take x(t) convolute with itself and then sample the obtained signal at Fs, if the sampled signal =delta[n], then Fs satisfies the Nyquist condition i.e: g(t)= x(t) (*) x