aliasing condition for radon transform?

Started by Anonymous in comp.dsp13 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: Interpolation

Started by robert bristow-johnson in comp.dsp12 years ago 9 replies

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the...

On Mar 31, 12:36 am, dbd wrote: > > When the sinc function is used as a prototype filter in windowed > filter design, the width of the sinc is scaled wider in time for > lowpass responses narrower than the Nyquist band. that's a choice that someone might make. > This would be done > for any lowpass anti-aliasing filter designed for sample rate > reduction with this me


Zero-padding, resolution and aliasing

Started by Amelia in comp.dsp12 years ago 16 replies

Hi, I understand that zero-padding a time domain signal cannot add any extra information to the spectrum, and that you only end up with a more...

Hi, I understand that zero-padding a time domain signal cannot add any extra information to the spectrum, and that you only end up with a more finely sampled spectrum. What I'm interested to know is whether zero-padding can cause spectral leakage? Or if I will always see a true representation of my original signal in the spectrum? Basically, I have been zero-padding my data by 100,000 poi...


dealing with aliasing

Started by fisico32 in comp.dsp10 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...


Anti-aliasing advice for de-interleaved signals

Started by PhilipOrr in comp.dsp10 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...


Power distribution across aliased signals

Started by starman3 in comp.dsp12 years ago 8 replies

I am having trouble understanding the effect of aliasing/imaging on the power of an output signal. Imagine I am using a DAC to generate a single...

I am having trouble understanding the effect of aliasing/imaging on the power of an output signal. Imagine I am using a DAC to generate a single frequency sine wave with frequency f well below fs/2 and with power P. Assume there is no reconstruction filter. So this frequency will image repeatedly across the frequency domain at n.fs-f and n.fs+f. I do not understand how the original power, P, i...


decimation and anti-aliasing

Started by prad in comp.dsp13 years ago 4 replies

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture...

Dear all, Thanks for all your replies but I am still a bit confused. Let me try rephrasing my question so that everyone gets a better picture of what I am trying to do. I have a large number (order of 10^13) of discretized data samples. I have to obtain the frequency spectrum of this data so that I can predict minima in the discretized data. I am using the FFTW C library to obtain the Frequency...


Aliasing of Quad-detector sensor signals

Started by Anonymous in comp.dsp2 years ago 8 replies

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...

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


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

Started by Don Pearce in comp.dsp13 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: window function-dsp

Started by Jerry Avins in comp.dsp10 years ago 1 reply

On Wednesday, January 19, 2011 2:25:07 AM UTC-5, rashmi venugopal wrote: > > Bharat: > > Total harmonic distortion can be measured in many...

On Wednesday, January 19, 2011 2:25:07 AM UTC-5, rashmi venugopal wrote: > > Bharat: > > Total harmonic distortion can be measured in many ways, but Rashmi asked > ab= > > out filter design. (A filter to suppress aliasing is needed even for your > F= > > FT approach.) > > > > Rashmi: > > Windowed-sinc design is historically interesting, but rarely the best > desig= > > n method when a


AM receiver - utilising bandpass undersampling and quadrature mixing

Started by thedspkid in comp.dsp14 years ago 1 reply

Hi, I am new to the whole DSP thing (very new) and have a question regarding the use of undersampling and quadrature mixing to demodulate a...

Hi, I am new to the whole DSP thing (very new) and have a question regarding the use of undersampling and quadrature mixing to demodulate a signal in a given range. I have been trying to research the topic but have not been getting very far. I understand how the bandpass signal can be undersampled at a rate of 2B without aliasing if the proper bandedge frequencies or chosen but have no real ...


Common practice - Aliasing in transition region?

Started by Anonymous in comp.dsp11 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 ...


Re: AM digital demodulation using the absolute value

Started by rickman in comp.dsp10 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...


A tricky problem

Started by Elnaz in comp.dsp8 years ago 1 reply

Hi all, I am convolving a vector of bits (zero-padded in between to avoid aliasing) with the erf function. On the paper, erf function goes...

Hi all, I am convolving a vector of bits (zero-padded in between to avoid aliasing) with the erf function. On the paper, erf function goes from - infinity to infinity which will cause erf functions in my convolutoin to cancel each other's tails out. However the problem is that in MATLAB no matter how long I define the function I will continue to have problem with the tails not canceling ea...


Is this 'clever' method of filtering legit?

Started by MartinC in comp.dsp14 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


OFDM/DFT/Sampling question.

Started by m26k9 in comp.dsp12 years ago 2 replies

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a...

Hello, I am pretty confused with some DFT/Sampling techniques and how these apply to OFDM. Fundamentally, if a (baseband) signal has a highest frequency component of f Hz, that signal needs to be sampled at 2f Hz for aliasing-free data reconstruction. My confusion begins with OFDM, which does the process in reverse. That is IDFT is performed first. So, in OFDM, the output of the IDFT blo...


Re: window function-dsp

Started by Jerry Avins in comp.dsp10 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


Anti-aliasing filtering for interleaved sampling approach

Started by Alexz in comp.dsp12 years ago 3 replies

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to...

Hello guys and girls, my first posting here. On some other, hardware design-related forum there was a question raised by someone whose goal is to built a sampling system to achieve 500 MHz sampling rate by means of two 250 Msps ADCs while driving them with samlping closk featured by 180 deg. phase shift. So that in time domain would be sampling by two channels in interleaved manner. Then the enti...


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

Started by Roy in comp.dsp11 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...


CAUTION! was "What is the advantage on high-sampling rate ?"

Started by Rick Lyons in comp.dsp16 years ago 150 replies

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web...

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web site http://www.lavryengineering.com/pdfs/sample.pdf you can see Dan's 1997 paper: "Sampling, Oversampling, Imaging and Aliasing - a basic tutorial". I recommend caution if you decide to read that paper. In the second paragraph Dan wrote: *** Sampling theory *** ...