how to decide a good sampling rate for sampling a function without obvious frequency?

Started by lucy in comp.dsp16 years ago 11 replies

Hi all, I am having trouble with my sampling problem: Basically I want to discretize a 2D Gaussian function f=gaussian(x, y) gaussian(x,...

Hi all, I am having trouble with my sampling problem: Basically I want to discretize a 2D Gaussian function f=gaussian(x, y) gaussian(x, y)=1/(2*pi*sigmax*sigmay)*exp(-0.5*(x^2/sigmax^2+y^2/sigmay^2)); In my experiments using Matlab, I am using square grids to deiscretize the above function. Suppose I have a grid -- [-N..N, -N..N] where N is the number of samples in one axis. So a...


Resampling and Convolution

Started by nazmat in comp.dsp14 years ago

Hi all, How can i convulve a data signal with a sampling rate of 100GHz with an impulse response of sampling interval of 0.167ns and the samples...

Hi all, How can i convulve a data signal with a sampling rate of 100GHz with an impulse response of sampling interval of 0.167ns and the samples are not evenly spaced.Thank you all. Nazmat


Reducing sampling rate, compiler optimization bug???

Started by bernd007 in comp.dsp13 years ago 6 replies

Hello, I use the PADK (Professional Audio Development Board) with a TI TMS320C6727 DSP and I want to reduce the sampling rate to 8kHz......

Hello, I use the PADK (Professional Audio Development Board) with a TI TMS320C6727 DSP and I want to reduce the sampling rate to 8kHz... The problem is, that the minimum possible sampling rate of the board is 32 kHz, so I have tried to solve the problem with a simple counter... I have an interrupt service routine (ISR), which is called with 32 kHz. In this routine I have a counter, counting to...


Something like cross correlation for the time domain

Started by in comp.dsp7 years ago 2 replies

From time to time I have to do cross correlations of signals with different sampling rates. The problem is that I do not know the ratio of...

From time to time I have to do cross correlations of signals with different sampling rates. The problem is that I do not know the ratio of the sampling rate, e.g. because longer recordings are taken with different crystal oscillators for the sampling rate. The idea is to pass two recordings to an algorithm and get the pitch and the delay as result. Strictly speaking the result is two d...


choosing a sampling rate lesser than nyquist rate(sub nyquist rate)

Started by praveen in comp.dsp17 years ago 17 replies

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60...

Hello, I have a signal consisting of 4 harmonics (200k,400k,600k,and 800k Hz) and dc component.The signal is very pure and SNR better than 60 dB. I have to sample it in sub Nyquisit rate(lesser than 1600k). What sampling rate should i choose so that there no alaising. I cannot chose higher sampling rate because of my hardware constraints. waiting for reply regards praveen


Sampling rate conversion

Started by Ting Liu in comp.dsp16 years ago 5 replies

Hi experts: Does any one know an easy way to convert the following two formats back and forth. mono 16bits sampling rate 44100hz PCM mono...

Hi experts: Does any one know an easy way to convert the following two formats back and forth. mono 16bits sampling rate 44100hz PCM mono 16bits sampling rate 8000hz PCM I have used ACM stream API to perform the conversion, but it produced poor sound quality (Alias occurred). I guess Microsoft's codecs are not good enough. :-( Is there any easy and fast way to accomplish this on M...


phase accuracy and sampling rate | PLL and FFT

Started by dralban in comp.dsp11 years ago

I read somewhere that if sampling two signals with frequency f and a sampling rate of N. Then the uncertainty in phase shift between the...

I read somewhere that if sampling two signals with frequency f and a sampling rate of N. Then the uncertainty in phase shift between the two signals will be 360*f/N. Someone know under which conditions this is true and how this uncertainty can be minimized? Next question is, suppose the signals contains harmonics with order n, will then the uncertainty in phase angle between the harmonic be 360...


Sampling Question

Started by Anonymous in comp.dsp12 years ago 9 replies

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60...

Suppose I take samples at once per minute for a data logger. Do I still need an anti-aliasing filter? The sampling freq would be 1/T where T=60 secs...so how is this possible? The values of the capacitors etc would be hugh. Of course I could over-sample1000s of times...is this the norm? K.


Sampling frequency correction by Interpolation/Decimation technique?

Started by nqh in comp.dsp13 years ago

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver,...

Dear all, In OFDM system, we difine T is the sampling period at the output of IFFT (at transmiter), T' is sampling period at receiver, SFO=(T'-T)/T At receiver, after CP removal, the (m,n)-th sample of the time-domain received signal is r'(m,n),(include SFO). How can we use Interpolation/decimation technique to correct the r'(m,n) (to get desied signal r(m.n) without SFO) Who can help m...


Sampling theorem for narrow band signals

Started by Anatol in comp.dsp13 years ago 15 replies

Hello, Could someone explain me please how the sampling theorem formula for narrow band signals is obtained. In the literature and on the...

Hello, Could someone explain me please how the sampling theorem formula for narrow band signals is obtained. In the literature and on the web one can find a good explanation of the sampling theorem of band limited signals, fs > = 2fmax. The explanation of the formula for narrow band signals fs > = 2fmax/k is not so intuitive and not very clear. Could you give me a link to the sampling


Hilbert transforms and sampling rate?

Started by MatthewA in comp.dsp2 years ago

Hi all, My knowledge of biquad filter coefficients isn't great but from what I've experienced, they change based on sampling rate. but I've...

Hi all, My knowledge of biquad filter coefficients isn't great but from what I've experienced, they change based on sampling rate. but I've bumped into these two examples of hilbert transforms online and neither takes sampling rate into account. Are they okay to use or are they just based on 44.1kHz? Thanks, Matt // hilbert quadrature filter hilbertBiquad(x0) { x1 = biq


Mixing with a 3000Hz tone @ 8000 sampling rate

Started by Shafik in comp.dsp15 years ago 10 replies

Hello folks, I think this problem was addressed before but it wasnt quite solved. I have a system running at 8000Hz sampling rate. I need to...

Hello folks, I think this problem was addressed before but it wasnt quite solved. I have a system running at 8000Hz sampling rate. I need to mix (multiply) my incoming sampling signal with a 3000Hz tone. The problem is, 3000 is not a nice division of 8000. Hence the samples produced dont quite generate a "clean" 3000Hz, and I end up getting artifacts. Stepping through a sine table at ...


min. sampling frequency of DTMF decoder

Started by Lionel Lewis in comp.dsp15 years ago 3 replies

Given the following frequencies in DTMF : lfg = [697 770 852 941]; % Low frequency group hfg = [1209 1336 1477]; % High frequency...

Given the following frequencies in DTMF : lfg = [697 770 852 941]; % Low frequency group hfg = [1209 1336 1477]; % High frequency group what is the min. sampling frequency and the number of samples required (using FFT) to detect the keys? I tried Nyquist sampling frequency (2*max freq = 2*1477 ~3000 ) and for the min. freq resolution requirement of 73Hz (from 697-770Hz) the num...


simple code for audio (or signal or sound ) sampling using gibbs sampling

Started by Anonymous in comp.dsp5 years ago 2 replies

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please...

hello everybody, I search in google a lot but I can not find code of using gibbs sampling in audio (signal or sound) processing. if you can please help me. thanks in advance


How do I evaluate the power spectra density of QAM signals?

Started by Franc in comp.dsp14 years ago 1 reply

My system comprises of a QAM mapping and up-sampling filter, now I need to evaluate the PSD of the I/Q data after up-sampling. How do I do that...

My system comprises of a QAM mapping and up-sampling filter, now I need to evaluate the PSD of the I/Q data after up-sampling. How do I do that in Matlab? TIA.


Sampling rate required to resolve separate sonar echoes

Started by Nicholas Kinar in comp.dsp12 years ago 27 replies

I am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to...

I am wondering if there is any criterion relating the ability to resolve the smallest time-duration echo to the sampling rate of an ADC used to convert a detected analog sonar signal to a digital sequence. The Nyquist sampling theorem states that you must sample a signal at greater than 2*f_h, where f_h is the maximum frequency of interest in the analog signal. A common mistake is to a...


Sampling Theorem history

Started by Tom in comp.dsp17 years ago 10 replies

It is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have...

It is generally credited that the Sampling Theorem is due to fistly the Mathematician Whittaker and Shannon and the Russian Kotelnikov. I have no doubt that Whittaker was first but was Shannon aware of Whittakers work? Also where does the Russian engineer fit in? What role did Nyquist play. We talk of the Nyquist frequency (half sampling) but why name it this if the work is due to Shannon? Or...


Speech sound from the web: Sampling rate, mono vs stereo?

Started by Anonymous in comp.dsp6 years ago 1 reply

I am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If...

I am capturing human speech on the web. I find there are examples that people are sampling at 44.1 and 48kHz. All of them are also stereo. If the sole purpose of capturing the sound is for extracting features, what might be the minimum or optimal sampling rate? Is there any value in Stereo signal? Am I correct that the left and right speech samples are identical in stereo - so just ignori...


need help in non uniform sampling

Started by ankit in comp.dsp9 years ago 5 replies

hello, im currently doing my masters.. i need to decide my topic of my thesis in DSP on non uniform sampling of signals.. your suggestions...

hello, im currently doing my masters.. i need to decide my topic of my thesis in DSP on non uniform sampling of signals.. your suggestions will be of great help... thanx


ADSP21364: 8kHz sampling rate?

Started by Nicholas in comp.dsp15 years ago 9 replies

Hello, I am running the "talkthru"-example on ADSP21364 EZ-KIT. The sampling rate is 48kHz. Is it possible to reconfigure the sampling...

Hello, I am running the "talkthru"-example on ADSP21364 EZ-KIT. The sampling rate is 48kHz. Is it possible to reconfigure the sampling rate to 8kHz? Thanks.