I'm not sure how to calculate aliasing components as I've only dealt with systems that are sampled above fs > = 2B. So if I want to make sure...

I'm not sure how to calculate aliasing components as I've only dealt with systems that are sampled above fs > = 2B. So if I want to make sure that I have aliasing but the components due to aliasing are at most 20dB below the maximum amplitude of the signal I don't know how to do that.

Hi folks, for those who are bored on this sunny winter afternoon, a riddle: Assume you are given a continuous but not necessarily...

Hi folks, for those who are bored on this sunny winter afternoon, a riddle: Assume you are given a continuous but not necessarily band-limited periodic function f, and some arbitrary time constant T (not related to the period of f). Is it possible to regularly sample f with sampling period T' > = T, such that f can be exactly reconstructed from the samples? Yes / No answers are not acc

A very basic question: Take two properly sampled signals (more than nyquist). Now I mulitply the two sample streams. From a continous time...

A very basic question: Take two properly sampled signals (more than nyquist). Now I mulitply the two sample streams. From a continous time view point it is easy to see that the product could have frequencies for which the initial sampling rate wouldnt be enough. So in any DSP system is one not supposed to multiply two signals? Has this something to do with whether multiplication with a v...

It's even more interesting to ponder this: Start with a set of sinusoidally weighted gray stripes going from peak black to peak white. This...

It's even more interesting to ponder this: Start with a set of sinusoidally weighted gray stripes going from peak black to peak white. This is a pure sinusoid plus a constant in one dimension. If we align the stripes either in X or in Y then we can have all these discussions about sampling and filtering as if it were all in 1-D. What happens if the stripes are rotated in XY by 45 degre...

Dear All I have some doubts regarding synchronisation. Please tell me if my understanding is right or not. Currently I am concerned abt...

Dear All I have some doubts regarding synchronisation. Please tell me if my understanding is right or not. Currently I am concerned abt carrier and symbol synchronisation. Now by symbol synchronisation, I understand that we need to sample the o/p of matched filer at right instant. NOw due to delay in channel, symbols (or samples per symbol) can arrive before or after the right sampling in...

Frequently Asked Questions (F.A.Q.)..... 1. (A Frequently Added Quotation, F.A.Q., is appended below.) Assuming that we were able to generate...

Frequently Asked Questions (F.A.Q.)..... 1. (A Frequently Added Quotation, F.A.Q., is appended below.) Assuming that we were able to generate a Diracian, and then produce a comb of them by delays and by superposition, there wouldn't be a factor of "T" in such superposition, so where does yours come from? Where did the factor of "T" come from in the "sampling function" in your opening li...

Hello all, When we process a signal, usually we represent it as a sequence of samples uniformly spaced in time. However if the signal can...

Hello all, When we process a signal, usually we represent it as a sequence of samples uniformly spaced in time. However if the signal can only take 0 or 1 values, the whole information is in the time of transitions. If we implement the processing as the interrupt on transitions instead of the regular sampling, the computing demand can be done less and the accuracy can be better. The i...

I need help with this situation that is stumping my intermediate DSP knowledge. We have an ADC sampling at 2.2GHz, with the data going into...

I need help with this situation that is stumping my intermediate DSP knowledge. We have an ADC sampling at 2.2GHz, with the data going into Virtex 6 FPGA. I need to take that data stream and downconvert and decimate it into a much slower, ~60 MHz channel that is down at, say, 50-110MHz. (there will actually be many 60MHz channels downconverted from the same wideband input data.) Conceptually I ...

I seem to remember from my DSP days at uni that there is a java website somewhere that is able to calculate discrete convolution coefficients...

I seem to remember from my DSP days at uni that there is a java website somewhere that is able to calculate discrete convolution coefficients for a few simple filter types based on the required cut-off and sampling frequency. Anybody know of such a site? I also have a copy of Maple which I'm sure would make light work of such an elementary task. Nathan

Kinda off topic -- A month or two ago there was a spate of postings on these groups displaying a profound misunderstanding of how to apply...

Kinda off topic -- A month or two ago there was a spate of postings on these groups displaying a profound misunderstanding of how to apply Nyquist's theorem to problems of setting sampling or designing anti-alias filters. I helped folks out as much as I could, but it really demands an article, which I am currently working on. The misconceptions that I noticed pretty much boiled down...

Hello all, I am struggling with what happens to the time axis in a sampled sequence when it is collapsed by adding every two samples...

Hello all, I am struggling with what happens to the time axis in a sampled sequence when it is collapsed by adding every two samples together, with or wihout averaging. I am working in the spatial domain but believe the principles should be the same as with time varying signals. I understand decimation and how it effectively resamples at a lower sampling frequency and has the potential ...

Hi, I was wondering if anyone could tell me what type of DSP i would need for a basic voice recognition system. The system would you a...

Hi, I was wondering if anyone could tell me what type of DSP i would need for a basic voice recognition system. The system would you a matched filter of 1000 samples and can have a sampling frequency of anywhere from 2200 toi 22050 Hz. Also the samples have a resolution of 16 bits. I'm confused on what sort of spec it would require e.g. in MIPS or MFLOPS. Any help would be very much apprie...

Hello. I have a sampled signal from a piece of scientific equipment where each sample is also accompanied by a time delta of extremely high...

Hello. I have a sampled signal from a piece of scientific equipment where each sample is also accompanied by a time delta of extremely high precision. The sample rate is as such anything but consistent. Needing to convert this into a more traditional form I need to low-pass this data before re-sampling. I could certainly oversample the signal and work from that, but since this is only a more...

Hi there, using waveshaping we can generate odd harmonics, right? Using some tricks like adding square we can have even harmonics. Is there a...

Hi there, using waveshaping we can generate odd harmonics, right? Using some tricks like adding square we can have even harmonics. Is there a way to get negative harmonics? E.g. Sampling rate 44100, someone feeds it with 1kHz, I want to get 500Hz, or 250Hz for example. I have no idea if it isn't completely useless, but it may be worth experimenting with :). Thanks in advance!

Recently I'm trying to solve a following problem: Let's say that we have some data that has been created by sampling some sinusoidal wave (ie....

Recently I'm trying to solve a following problem: Let's say that we have some data that has been created by sampling some sinusoidal wave (ie. a wave that "looks" like a sine wave in which frequency is changing over time in some unknown manner - it is still a continuous function, has values between -1 and 1, but sometimes it is "squashed", othertimes it is "widened"). What's more, our samp...

Hi all. I have this application where I have several minutes of acoustic recordings at a few tens of kHz sampling frequency. I need to...

Hi all. I have this application where I have several minutes of acoustic recordings at a few tens of kHz sampling frequency. I need to compute power spectra from these data with confidence intervals. I use a one-pass variance estimator (pseudo code): %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% X1=3D 0; % Nfft x 1 data vectors X2=3D 0; for k=3D1:K Xk=3...

Frequently Asked Questions (F.A.Q.)..... 1. (A Frequently Added Quotation, F.A.Q., is appended below.) Assuming that we were able to generate...

Frequently Asked Questions (F.A.Q.)..... 1. (A Frequently Added Quotation, F.A.Q., is appended below.) Assuming that we were able to generate a Diracian, and then produce a comb of them by delays and by superposition, there wouldn't be a factor of "T" in such superposition, so where does yours come from? Where did the factor of "T" come from in the "sampling function" in your opening li...

Hello, I am designing a sample rate converter from 8khz to 8.4 khz, which gives an upsampling rate of 21 and downsampling rate of 20. I...

Hello, I am designing a sample rate converter from 8khz to 8.4 khz, which gives an upsampling rate of 21 and downsampling rate of 20. I am implementing it using polyphase filters. I have got a filter with cut-off frequency of 21. This means I have 21 filters each of them shifted and upsampled by 21. And finally I cascade the output and downsample by 20. is this the right way of doing ...

Hi, I am very new to the subject. I spent hours online on trying to find how to use autocorrelation and fft in matlab, in order to separate...

Hi, I am very new to the subject. I spent hours online on trying to find how to use autocorrelation and fft in matlab, in order to separate a signal from noise. I have this so far: t = 0:.01:1; % independent (time) variable A = 8; % amplitude Fs = 101; % Sampling frequency frequency1 = 2; frequency2 = 12; sine...

HEllo, I have an acquisition system of 30 measurements. The sampling rate for one acquisition is 1000 Hz. Each acquisition must be filtered...

HEllo, I have an acquisition system of 30 measurements. The sampling rate for one acquisition is 1000 Hz. Each acquisition must be filtered with a FIR with 16 Taps and 16 sample of the signal acquistion will be used. I will use a TSM320C2xxx with a frequency of 150 Mhz. How many MIPs do i need to achieve thoses filtering operations. Thank oyu