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

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

Hello I need help, in DFT, the frequency domain?s independent variable can be refereed to in many ways, one way is being a fraction of the...

Hello I need help, in DFT, the frequency domain?s independent variable can be refereed to in many ways, one way is being a fraction of the sampling rate. could some one please explain why X[ ] independent variable runs between 0 and 0.5, if you say because discrete data can only contain frequencies between DC and ? the sampling rate, then if the sampling rate (x[ ] has) 128 points the...

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.

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

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32...

hi every body, i am using 6713DSK for my project.i use AIC23 for sampling a 1khz sin wave and collect them in a buffer.i configured MCBSP for 32 bit one frame format.after sampling i shift 32 bits samples to right to just have 16 bits of left channel.then when i plot the collected samples in CCS it is not a 1khz sin wave.does anyone know what is my problem. i dont have mush time.please help me if...

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

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 ...

Gents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling...

Gents, I know that the amount of system phase lag, introduced by a DSP alone, decreases with an increase in its sampling frequency. Assuming the ADC read + DAC write both happen within a sample period and assuming the only work the DSP does is setting its output=input, does anyone know a formula that describes the relationship between phase lag and sampling is linear? I set up a small...

Hi, 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...

Hi, 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 MS/s. 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 a cosnius at 60 MS/s and I need to multiply theses sinus, cosinus with the 15 MHz IF2. But the...

Hi Guys, I want to have a band pass filter with cutoffs 0.15Hz and 2Hz and also a low pass filter with a cutoff of 10Hz. I want to choose...

Hi Guys, I want to have a band pass filter with cutoffs 0.15Hz and 2Hz and also a low pass filter with a cutoff of 10Hz. I want to choose either one at a time. I have to give the sampling frequency to both filters and so if I give the same sampling frequency, is it not going to be a problem? How do we decide which sampling frequency works well for both filters? Please give me some info ove...

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...

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...

I have some questions regarding the sampling frequency of a digital filter? This is the rate at which the internal stages are clocked, why does...

I have some questions regarding the sampling frequency of a digital filter? This is the rate at which the internal stages are clocked, why does this have to be twice the incoming data rate. I don't understand where Nyquist comes in the filter design.Our incmoing data was generated by sampling a analog at twice frequency, and now my filter must run at twice the data rate.Why?

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than...

I (vaguely) heard that sampling complex-valued data does not abide by the Nyquist rate criteria, i.e., the sampling rate fs can go lower than Nyquist rate and it still can avoid aliasing and reconstruct perfectly... Is that true? Any theory behind it? Thanks a lot

Hi, I'm new to OFDM. I read in several textbooks and papers that timing synchronization can be separated into symbol synchronization and...

Hi, I'm new to OFDM. I read in several textbooks and papers that timing synchronization can be separated into symbol synchronization and sampling clock synchronization. The latter can be further broken down into sampling phase offset and sampling frequency offset. There is also carrier frequency synchronization (i.e., carrier frequency recovery technique), which is utilized to eliminate the...

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...

This is something I noticed a short while back which I've not seen explicitly mentioned as such. The Wigner distribution, when applied to the...

This is something I noticed a short while back which I've not seen explicitly mentioned as such. The Wigner distribution, when applied to the analytic signal associated with a real signal, has a 1/2-continuous 1/2-discrete sampling grid built into it that is scale invariant and hyperbolic, much like the sampling grid for the discrete wavelet transform; and it can be reproduced by its sampl...

Hello; I wrote this basic demo to help me learn about DSP for a class (digital filtering) at my school that I am taking now, which...

Hello; I wrote this basic demo to help me learn about DSP for a class (digital filtering) at my school that I am taking now, which illustrates sampling a signal, then it generates the CTFT (cont. time Fourier transform) and then the DTFT (discrete time Fourier transform) from the samples. Both the magnitude and the phase are plotted. One can adjust the sampling frequency, signal dura...

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g....

Hi, Is anyone using equipment like a VSA for data capture? Any idea how I can set the sampling rate in a Tektronix RSA? Whatever I set (e.g. using SENSe:ACQuisition:BANDwidth or SENSe:IQVTime:FREQuency:SPAN), the sampling rate is always 150 MHz! Frustrating... With an Agilent MSG I am transmitting an LTE signal which is sampled at 30.72MHz and to be received by the Tektronix VSA. (But ...