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Datasheet Question

Started by Mauritz Jameson in comp.dsp5 minutes ago 14 replies

Section 7.2.1 on page 11 in this PDF: http://cache.freescale.com/files/sensors/doc/app_note/AN4076.pdf describes how to convert a 14-bit...

Section 7.2.1 on page 11 in this PDF: http://cache.freescale.com/files/sensors/doc/app_note/AN4076.pdf describes how to convert a 14-bit value into a decimal fraction, but I'm not sure I understand it so I'm hoping that somebody in this group could take a look at the PDF and tell me if my interpretation (see below) is correct? 1. Read in first byte (8 most significant bits) denoted MSB ...


Minimization / maximization problem

Started by Piergiorgio Sartor in comp.dsp38 minutes ago 2 replies

Hi all, I need some hint or pointer in order to try to solve a problem. This might not be the right place, but since here wise people are,...

Hi all, I need some hint or pointer in order to try to solve a problem. This might not be the right place, but since here wise people are, maybe I got lucky (I tried sci.math too). If you know an other possible NG, please let me know. I've a set of vectors, same size, these are "samples" of different populations, of course I've by far more vectors than populations. These (column) vect...


zero padding and periodic waveform energy calculation

Started by helloworld in comp.dsp39 minutes ago 1 reply

Consider a 100 Hz waveform sampled at 10kHz. One period would consist of 100 samples. I'm trying to better understand the energy of this...

Consider a 100 Hz waveform sampled at 10kHz. One period would consist of 100 samples. I'm trying to better understand the energy of this waveform and the same waveform with 28 zeros. I'm asking because I see two equations pertaining to the energy of a periodic waveform. (1) is the summation of squares (2) same as 1 with the addition of multiplying the sum by the duration of the wavefor...


Discrete -Time Fourier Transform

Started by Shadow Flux in comp.dsp2 hours ago

So from what I searched in the net, DTFT is the continuous Fourier transform of a discrete (or quantized?) signal while DFT is the...

So from what I searched in the net, DTFT is the continuous Fourier transform of a discrete (or quantized?) signal while DFT is the discrete Fourier transform of the quantized signal. Meaning DFT is used in actual Systems since it gives out data which can be processed by computer. If that is the case, what is DTFT for? Is it used in actual systems or just used for mathematical proving and analysis?...


FM SW Demodulated Audio - I get just noise

Started by b2508 in comp.dsp4 hours ago 9 replies

Hi everyone, this is my first post and it is quite important and urgent to me. I am new with this area so forgive me if I have some gaps in...

Hi everyone, this is my first post and it is quite important and urgent to me. I am new with this area so forgive me if I have some gaps in knowledge. I am trying to receive FM radio station (possibly the one with only voice) and play it on speakers. This is all done in Labview (thats mandatory), first NI board is used to downconvert 50 MHz of bandwidth from desired center frequency to...


FIR filter bandpass filter from lowpass filter prototype

Started by hyuckin kwon in comp.dsp10 hours ago 4 replies

I want to make bandpass filter from lowpass filter prototype. first i made lowpass filter and frequency translated using cosine function and...

I want to make bandpass filter from lowpass filter prototype. first i made lowpass filter and frequency translated using cosine function and i saw the freq. response of translated coefficient. but it differs from lowpass filter response. b = firceqrip(368, 4.3/(122.88/2), [0.1, 0.0018], 'slope', 0); fvtool(b); n=0:length(b)-1; cos_b = b.*(cos(((3*pi)/8)*n)); fvtool(cos_b) > lowp


Basic Sampling Theory Question

Started by old_ee in comp.dsp14 hours ago 36 replies

Hi, If I have a sine wave with period T. The sampling theory says I can recover it by sampling at T/2. That is two points, barely enough to...

Hi, If I have a sine wave with period T. The sampling theory says I can recover it by sampling at T/2. That is two points, barely enough to draw a straight line. What am I missing here? --------------------------------------- Posted through http://www.DSPRelated.com


Analog error correction codes

Started by Anonymous in comp.dsp20 hours ago 6 replies

There are 2 versions of analog error correction codes that I found. One is from the biological sciences:...

There are 2 versions of analog error correction codes that I found. One is from the biological sciences: https://clm.utexas.edu/fietelab/Papers/nn.2901.pdf It uses simple mapping onto phase space to do error correction. The second uses chaos theory to expand the state space: http://arxiv.org/abs/1105.1561 Neither of them give a really clear description of the decoding a


FIR filter bandpass filter from lowpass filter prototype

Started by tommy_kwon in comp.dsp1 day ago 1 reply

I want to make bandpass filter from lowpass filter prototype. first i made lowpass filter and frequency translated using cosine function and...

I want to make bandpass filter from lowpass filter prototype. first i made lowpass filter and frequency translated using cosine function and i saw the freq. response of translated coefficient. but it differs from lowpass filter response. b = firceqrip(368, 4.3/(122.88/2), [0.1, 0.0018], 'slope', 0); fvtool(b); n=0:length(b)-1; cos_b = b.*(cos(((3*pi)/8)*n)); fvtool(cos_b) > lowp


Reference level in RF receiving instruments

Started by b2508 in comp.dsp1 day ago

Hi all, I was wondering if someone could explain me or point me in the right direction (some pdf, tutorial or similar) on how to choose...

Hi all, I was wondering if someone could explain me or point me in the right direction (some pdf, tutorial or similar) on how to choose Reference Level for Spectrum Analyzer or VSA and how do I actually know that it is good or bad. I am aware that I should probably maximize the signal by setting reference level a bit above the signal and not too much but is there more to it than that? A...


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