Getting better FFT resolution in lowend by downsampling?

Started by jungledmnc in comp.dsp8 years ago 5 replies

Hi, just a little idea that came in my seriously distorted mind :), hopefully it's not a complete nonsense. When analysing audio signals, we...

Hi, just a little idea that came in my seriously distorted mind :), hopefully it's not a complete nonsense. When analysing audio signals, we like log domain, so Q transform, but AFAIK it hasn't been implemented fast enough yet, right? So I thought we can just use FFT and then use another FFT specifically for low-frequencies. Example: SR 44kHz, nyquist 22kHz, FFT 8192, resolution 22000/409...


SINE SQUARE NYQUIST

Started by arch...@hotmail.com in comp.dsp14 years ago 9 replies

TO sample and be able to reconstruct the sine wave we need to sample it at > 2 x maximum frequency present. Can the same sampling rate be used...

TO sample and be able to reconstruct the sine wave we need to sample it at > 2 x maximum frequency present. Can the same sampling rate be used to sample a square wave of the same frequence ? thanks, ajw


Newbie ? about FFT Bins and Resolution BW

Started by Paul in comp.dsp11 years ago 1 reply

If i understand this correctly, the inverse of the sample time will give us the frequency resolution, which is the same as the resolution...

If i understand this correctly, the inverse of the sample time will give us the frequency resolution, which is the same as the resolution bandwidth on a spectrum analyzer, right? So if I'm going into an ADC, and i want the receive KTB to stay 14dB or so above the thermal and quantization noise of the ADC, then i can sample at 100MHz, and the band width can be at the Nyquist, or 50MHz. So ...


Sampling: What Nyquist Didn't Say, and What to Do About It

Started by Tim Wescott in comp.dsp8 years ago 135 replies

I know there's a few people out there who actually read the papers that I post on my web site. I also know that the papers have gotten a bit...

I know there's a few people out there who actually read the papers that I post on my web site. I also know that the papers have gotten a bit ragged, and that I haven't been maintaining them. So here: I've made a start. http://www.wescottdesign.com/articles/Sampling/sampling.pdf My intent (with apologies to all of you with dial-up), is to convert the ratty HTML documents to pdf as...


FFT build with DSP

Started by Eugene in comp.dsp14 years ago 2 replies

Hello Shawn Steenhagen, I didn't understand you message, you wrote: " The output of the rfft( ) is a complex result in the form: y(0)Re...

Hello Shawn Steenhagen, I didn't understand you message, you wrote: " The output of the rfft( ) is a complex result in the form: y(0)Re y(nx/2)Im (DC and Nyquist) y(1)Re y(1)Im y(2)Re y(2)Im etc. " what does it mean? If I use rfft(), then I thoght, I should get a real numbers of fft. after use rfft() I have one array with numbers of result. if I want to show it in "code comp...


Nyquist and the Complex-Real FFT

Started by vanlansl in comp.dsp11 years ago 6 replies

I've heard that there is a way to essentially double the sampling rate of a complex input signal into a real FFT by alternating the real and...

I've heard that there is a way to essentially double the sampling rate of a complex input signal into a real FFT by alternating the real and imaginary signals in time (given that the I and Q are sampled independently)? Does this really work? If it does, are there only certain conditions (relationships between sample time and frequency) where it does? Thanks Much!


Digitally compensating a low-pass (integrator)

Started by Peter Mairhofer in comp.dsp8 years ago 8 replies

Hi, I have a signal f(t) with nyquist rate W, i.e. the maximum frequency is W/2. This signal is filtered with an integrator (simulated in...

Hi, I have a signal f(t) with nyquist rate W, i.e. the maximum frequency is W/2. This signal is filtered with an integrator (simulated in Simulink) the following way: f_I(t) = \int_t^{t+1/W} f(t) dt In words: I integrate the signal for a period of 1/W, then the integrator is reset. It is obvious that the signal won't be the same afterwards; however I only integrate for 1/nyquistr...


convert sum of two sines to square wave?

Started by Funky in comp.dsp15 years ago 24 replies

Suppose two 16 bit words define the frequency of two sine waves from 0.1Hz to 20 Khz +/-0.1Hz which are then summed, amplified by "infinity" and...

Suppose two 16 bit words define the frequency of two sine waves from 0.1Hz to 20 Khz +/-0.1Hz which are then summed, amplified by "infinity" and then clipped, giving a pulsed output. In theory, I think this could be done using a DSP to do everything, but what's the best strategy? Method 1: 1. Store a Nyquist sampled sinewave in Eprom 2. upsample the lower frequency to that of the higher fr...


how to adjust poles and zeros for reducing error

Started by alpha1 in comp.dsp4 years ago 13 replies

hi, I am designing a deemphasis filter of 15/50us specification. I am using using bilinear transform method. But at frequencies near to...

hi, I am designing a deemphasis filter of 15/50us specification. I am using using bilinear transform method. But at frequencies near to nyquist frequency I am getting an error of 1dB. I tried pre wrapping. But I gives error reduction only at the frequency where the wrapping is applied. So I would like go for pole zero adjustment to reduce the error. What are the guidelines that I should fol...


Singer Acceleration Model and Kalman

Started by nicosd in comp.dsp14 years ago 1 reply

Hi, I'm looking for someone familiar with the Singer acceleration model. Could such a person explain to me why Singer makes no assumption...

Hi, I'm looking for someone familiar with the Singer acceleration model. Could such a person explain to me why Singer makes no assumption on the sampling rate and the nyquist theorem? How can he say things such as the correlation coefficient goes to infinity, when it is actually bounded by 1/2T where T is the sampling period? This is driving me bananas!! thnaks This message was se...


Sampling Complex Signal

Started by Walter Wego in comp.dsp7 years ago 22 replies

Hi, I know you have to sample above Nyquist or you risk aliases folding into the passband. But it if your signal is complex, it seems to me...

Hi, I know you have to sample above Nyquist or you risk aliases folding into the passband. But it if your signal is complex, it seems to me that in a given interval of time you?ve collected twice the information you would have for a real signal, or that you?ve effectively doubled your sample rate. I got that same message when looking through old posts on DSP Related. I thought that


newbie: fir filter design

Started by Anonymous in comp.dsp14 years ago 17 replies

Hi again, I need to design a FIR filter with an even number of coefficients. I was wondering if the following formula is correct for...

Hi again, I need to design a FIR filter with an even number of coefficients. I was wondering if the following formula is correct for calculating the coefficients for a lowpass filter with hamming window - for(n=0;n


Related to FIR filters

Started by I. R. Khan in comp.dsp14 years ago 20 replies

Hi, I am comparing a maximally linear (ML) FIR differentiator with an equiripple one. The ML filter has a monotonic response, and comparing...

Hi, I am comparing a maximally linear (ML) FIR differentiator with an equiripple one. The ML filter has a monotonic response, and comparing the relative error curves (1- actual / ideal magnitude response) of both filters, it is found that ML is more accurate in the lower band below 0.88 * nyquist. Can I say that if we set the sampling frquency beyond (2/0.88) * maximum frequency in th...


Shannon and negative frequencies

Started by Chris Bore in comp.dsp9 years ago 4 replies

In Shannon's paper that sets out the Sampling Theorem: http://www.stanford.edu/class/ee104/shannonpaper.pdf he formally states (using B...

In Shannon's paper that sets out the Sampling Theorem: http://www.stanford.edu/class/ee104/shannonpaper.pdf he formally states (using B to represent the Nyquist frequency): "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart." "A similar result is true if the band d...


Re: Interpolation

Started by robert bristow-johnson in comp.dsp11 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


Nyquist and rectangular waveforms

Started by Bob Masta in comp.dsp5 years ago 23 replies

This 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...


Do Nyquist/filtering requirments hold for raster video digitizing?

Started by Jeff Miller in comp.dsp14 years ago 16 replies

Hi, I'm trying to convert my electron microscopes for display on a PC rather than the awkward and obsolete consoles. I recently bought an NI...

Hi, I'm trying to convert my electron microscopes for display on a PC rather than the awkward and obsolete consoles. I recently bought an NI IMAQ-1422 (ebay of course) to do the higher level framing and rasterizing. I'll use either a delta-sigma or sampling SAR 16 bit AD converter for the digitizing and a pair of DACS driven by count-down timers to scan the beam. Settling time aside, ...


Adding noise to ADC input to increase accuracy

Started by dwjbosman in comp.dsp8 years ago 2 replies

Dear all, I have just been reading about the use of oversampling in a decimator to enhance the resolution of a sampled signal. The idea is...

Dear all, I have just been reading about the use of oversampling in a decimator to enhance the resolution of a sampled signal. The idea is to sample a signal containing 0.5LSB white noise at samplerate of N times the Nyquist rate. It is then possible to extract log4(N) extra bits of resolution by summing N samples. However there is another scheme which can provide much more of an increase...


A/D Practical Aperture Limit

Started by Jon Mcleod in comp.dsp11 years ago 11 replies

With current A/D technology, what is the upper limit of what can be directly sampled? Super-nyquist is OK -- I'm wondering about the highest...

With current A/D technology, what is the upper limit of what can be directly sampled? Super-nyquist is OK -- I'm wondering about the highest absolute frequency that can be directly sampled in any practical sense. Could I directly sample a 900MHz GSM antenna? Or a 1.8G CDMA antenna? How high can you go?


Frequency offset vs. Symbol Rate Problem

Started by Frank Cassidy in comp.dsp14 years ago 3 replies

Hi Everybody, I have a problem that I thought would be pretty common, but I haven't found a good solution yet. The problem is basically this:...

Hi Everybody, I have a problem that I thought would be pretty common, but I haven't found a good solution yet. The problem is basically this: I have to accommodate a frequency offset of 1450Hz on a signal coming in at 1200sym/sec. If I sample 6 times a symbol, I get the required bandwidth to satisfy Nyquist's, but the correlator length is too much for the dsp to handle. Does anybody ha...