As a result of a measurement I obtain a bandlimited signal centered at
144 kHz (12kHz Bandwidth). I was planning to use undersampling
(Fs=192kHz) to fold this signal to 48kHz in the digital domain.
I am having a difficulty to find an ADC for this purpose. The only
ADCs I could find (I...
I would like to undersample a 70 MHz IF signal at a sampling frequency
of 55 MS/s. So this undersampling method creates a 15 MHz IF2 whith 55
Now I would like to donwconverter this 15 MHz IF2 to baseband (0 IF)
with quadrature sampling, so I need to generate a sinus at 60 MS/s and...
I'm looking for a software to compute the frequency sampling for
undersampling an IF radio spectrum.
For example, if I have a IF of 455 kHz with 80 kHz bandwith, I can
sample at 200 kHz and the second IF would be 55 kHz.
I have try to compute with software designed for RF mixer but it...
Hi! I want to make an ultrasonic leak detector. Generally, they consist
mainly of an ultrasonic transducer and an analog demodulator with a
carrier of 40Khz which they enable to hear and to measure an audio signal
from 0 to 1.5 or 3Khz.
How I can make that digitally? I'm thinking about to make...
If you undersample, then in fact you do need an analog hold circuit
that is fast enough for the total carrier frequency (2G) - but your ADC
need only be fast enough for the band you are in fact measuring.
This is because when undersampling, once the signal is held (the
expensive part) you have...
Hi sampling gurus,
I'm thinking about undersampling a 150Hz AM signal modulated to a 330MHz ca=
rrier, thus the signal bandwidth is 300Hz. I found an ADC has 600MHz analog=
ue bandwidth, and minimum Fs to be 5.1kHz. Suppose the bandpass filter is s=
harp enough for my my signal at 330MHz, if I...
If we change our wording of Nyquist requirement from minimum samples per
cycle to "its original, I believe" of minimum sampling frequency then I
wouldn't worry about number of samples per cycle. Sampling frequency stays
same whether you have one channel(I) or a pair(I/Q) representing the same
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...
> am i correct in thinking that if i band pass filtered the stream between
> 30 - 80 kHz, then took every eighth sample the audio would be aliased into
> the audible range?
No, undersampling is not the nature of your problem. You wish to scale
one base band down to f...
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
I have been trying to research the topic but have not been getting very
far. I understand how the bandpass signal can be undersam...
"Tom" wrote in message news: ...
> I have been reasing a paper that says that for acoustic beamformers (I
> suppose the same applies to EM beamformers), the distance between
> microphones must be
> where Lmin is the minimum wavelength.For 8kHz sampling frequency this makes
I have two 70 MHz signals (one a clean reference sine wave, the other
a lot less so), but wish to find the phase difference between these
The A/D, the AD9226:
has a maximum sampling rate of 65 MSPS, but is...
> If anyone is still interested:
> What I have done to improve the noise level is sample at 16 times the
> switching frequency, to give me 8 samples per switch state (where
> previously I had only a single sample).
> Then, I use the mean of these 8 to get the dif...
I'm new to DSP and also this group.
My question is "what can be the negative effect of Oversamling in terms of
Destortion in original signal?". As in undersampling two spectrum are
overlapped and we may not be able to get original signal.. is that any
kind of effect in oversampling?
On Sat, 14 Jan 2006 01:21:15 +0000, Vladimir Vassilevsky wrote:
Cool. Thanks, Vladimir.
You mentioned IC solutions that have the downsampling built in. Do you
have any specific suggestions that you believe would be a good choice for
my problem space?
I have just been doing some test and I noticed a concerning problem that I
may be having with the undersampling design (for FM demod).
I basically set up my board so that the adc input (which is clocked at
80MHz) feeds directly into a DAC also clocked at 80MHz.
If I put a 20MHz tone...
I think this is not so much about signal processing but about statistics:
When you say you are "undersampling" I agree: It is not possible to
reconstruct the actual "waveform" from the samples without loss of
Still, statistics will be able to extract useful information fro...
Jerry Avins wrote:
> In theory only. To resolve a signal at 2.00000X would require 1,000,000
> seconds. (OK: maybe only 150 hours.)
150 hours? Why?
> Right or wrong, that's what I meant.* What's more, to resolve Fs/2 - 10
> Hz, you also need to to sample for a time in the order of 1...
Read this astonishing story at
Here it is an excerpt:
A STORY ABOUT UNDERSAMPLING
by Angelo Ricotta - Rome, Italy
In the article "Turning Nyquist upside down by undersampl...
I was wondering if you could help me with this problem. I have an
accelerometer with a resonance of say 15kHz. I need to do envelope
detection of this (acts a bit like AM with 15kHz as the 'carrier'). The
bandwidth of this signal is lets say 4kHz so that we bandlimit from
> No - as Jerry has already explained, one of the important things that
> limits undersampling is the BW of the front-end of the ADC. Usually the
> analog front-end of an ADC will be several tens of times (say 10x) the max
> sampling rate it can support.
If the ADC ...
I'm wondering if any of you attended the Embedded
Systems Conference (also called "electronicaUSA") in
San Francisco last month.
I have a question about a paper that may be on the
CD distributed at the conference. The lecture paper
that interests me is titled: "Unde...
Sai Kumar wrote:
> Dear All,
> I have one problem in real time signal processing. My signal frequency is
> 4000Hz and the sampling frequency is 8000Hz. Actually it should be more than
> Becasue of that i am loosing the zero resting points of the sampled signal.
On Apr 27, 12:23 pm, Jerry Avins wrote:
> It needs to be clear that
> one can't decimate first and filter later.
I think there may be times where you can decimate first and filter. If
the significant components of the signal don't alias to be on top of
one another, a pers...
Rune Allnor wrote:
> I'm testing out various ideas that rely on the unwrapped phase.
> The problem is that the various unwrapping schemes apparently
> have their deficiencies, so I'd like a synthetic reference where
> I know the "true" answer (total phase), and then test the un...
When digitally mixing an IF down to baseband, one is left with a spectrum
that consists of the baseband (Fif - Fmix = 0Hz) and an image (Fif + Fmix).
If the IF is greater than the Nyquist freq, the image will wrap back into
the first Nyquist zone (0 to Fn).
Normally the next step in demodulati...
On Nov 5, 9:54=A0am, "alexgaas" wrote:
> Hi guys!
> Does anyone know any algorithm used in digital storage oscilloscope?
> I know that many DSOs use equivalent time sampling technique but what
> about the digital hardware (or algorithms) to filter and reconstruct the
> If a...
I am curious why such a system does NOT work. I am using a 16-bit
fixed-point DSP chip:
voice1 -> ADC -> DSP -> DAC -> voice2
I collect an array of samples and pass them through an FFT then an
inverse FFT. The sample size is 256 but obviously, that can be changed.
I have a problem with my fftw complex transform (1d) when I want to
undersample my signal.
I start with initial signal (Dimension 16384points), and get the FFT of
this signal in complex with fftw_plan_dft_1d (result is a 16384points
dimension too). This signal has 128 first non-zero va...
I am on a team developing a system that will record and process large
(1 GHz) bandwidth pulses. Doing some research I have came across some
one that suggested sampling in the 2nd nyquist zone, for two main
reasons. Sampling the signal at a higher frequency makes our analog
front end much bette...
Richard Owlett wrote:
> glen herrmannsfeldt wrote:
> > Jerry Avins wrote:
> > (snip)
> > > The question is whether imaginary numbers and negative frequencies
> > > are inherent qualities of the real world, or whether they are
> > > scaffolding for the simplified mathematics that ...
I'm fairly new to DSP and am trying to see the effect of undersampling on
bandwidth limited white noise in Simulink.
I took the bandwidth limited white noise source (set the BW to 1kHz), put
it through a zero order hold with Sampling rate set to initally 5kHz and
the output of this is connected t...
On 2/7/12 12:55 PM, Tim Wescott wrote:
> On Tue, 07 Feb 2012 17:27:21 +0000, Eric Jacobsen wrote:
> > Tim and I aren't the only people who casually refer to images as
> > aliases.
> > But, yes, I think "image" is a better term for the artifacts in
> > reconstructi...
On May 13, 8:02=A0pm, "bos1234" wrote:
> Consider the system below. At point B it is graphed. The question is what
> is the value of the signal at 15Khz. And what is the aliasing error at th=
> Why did they minus F_s-f=3D45 in the solution. And what is the value of t=...
Graeme Zimmer wrote:
> I have a nice little DSP radio working, see
> It uses two mixers to develop the quadrature data streams.
> I figured that I could eliminate the quadrature mixers,
> if I could ...
Imagine a thing you can get is an old oscilloscope where the timebase is unknown.
Imagine you want to know the phase between two signals.
Hence you are an engineer knowing the phase without knowing the delay ...
What I want to say:
Possibly there are algorithms calculating phase ...
murselonder wrote in news:1181921715.561258.174770
> 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 o...
I'm basically trying to design a power spectrum analyser that can
capture channels with a bandwidth of 100 MHz. I select either 10 MHz
- 110 Mhz or 135 MHz - 235 MHz and am sampling with a 250 MHz ADC.
The ADC has a full power bandwidth of 1 GHz and has 12 bits.
My goal is to be abl...
Hi, I'm trying to pick up some more DSP understanding and I've been using
Richard Lyon's book, Understanding Digital Signal Processing. Its been
very helpful thus far, but there's one example that's driving me crazy.
I've bounced this off another couple people here, but they seem confused
"Tim Wescott" wrote in message
> Chi Chian wrote:
> > i was thinking of a high speed (~250Msps) high resolution
> > (say> 15bits) ADC. is there such a device in the market?
> > how abt multiplexing a few slower but high resolution