aliasing condition for radon transform?

Started by Anonymous in comp.dsp12 years ago 5 replies

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...

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

Started by Rune Allnor in comp.dsp12 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


Is this 'clever' method of filtering legit?

Started by MartinC in comp.dsp12 years ago 12 replies

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...

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.

Started by Don Pearce in comp.dsp12 years ago 290 replies

On 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

Started by rickman in comp.dsp8 years ago

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?

Started by Ben Jackson in comp.dsp13 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


Re: window function-dsp

Started by Jerry Avins in comp.dsp8 years ago 1 reply

Bharat: 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

Started by qaisar in comp.dsp14 years ago 2 replies

Hello, 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

Started by vasindagi in comp.dsp11 years ago 28 replies

Hi, 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

Started by Shafik in comp.dsp15 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 ...


Anti-Aliasing filter

Started by naebad in comp.dsp14 years ago 21 replies

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...

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

Started by PhilipOrr in comp.dsp9 years ago 36 replies

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...

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

Started by Kamil in comp.dsp11 years ago 5 replies

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...

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?

Started by Funky in comp.dsp15 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...


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

Started by Roy in comp.dsp10 years ago 1 reply

Hi, 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.

Started by Qian...@gmail.com in comp.dsp11 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


dealing with aliasing

Started by fisico32 in comp.dsp9 years ago 3 replies

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...

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?

Started by Anonymous in comp.dsp10 years ago 9 replies

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...

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.

Started by A.E lover in comp.dsp12 years ago 1 reply

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...

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


sinusoids and aliasing...

Started by fisico32 in comp.dsp9 years ago 5 replies

Hello forum, while a composite signal (made of many sinusoids), if sampled at a sampling frequency f_s at least twice the largest frequency in...

Hello forum, while a composite signal (made of many sinusoids), if sampled at a sampling frequency f_s at least twice the largest frequency in the signal, can be "uniquely" reconstructed from its samples, a continuous pure sinusoid of freq f instead, no matter if sampled at twice or more its frequency, will give samples that can be the samples of other sinuosids, all those with frequency f+-n...