I am simulationg an OFDM system in Matlab for which I need to generate a sampling clock offset in my transmitted signal so that I can write an...

I am simulationg an OFDM system in Matlab for which I need to generate a sampling clock offset in my transmitted signal so that I can write an algorithm at the receiver to correct it. Can any body tell me how to generate a sampling clock offset in the Transmitted signal. I think I have to use interp1 function but I dont have a clear idea. Sampling Clock offset should be a linear function o...

Why use multiple sampling rates? The most important reason is for efficiency of calculation and/or memory. But what is the lowest sampling rate...

Why use multiple sampling rates? The most important reason is for efficiency of calculation and/or memory. But what is the lowest sampling rate possible? Well, the Nyquist theorem tells us that the sampling rate must be at least twice the bandwidth of the signal. Therefore, for maximum efficiency, you would like to do DSP processing at a rate which is a small multiple of the signal bandwidth. ...

(how) does coherent sampling (used primarily in A/D testing?) differ from I/Q demodulation? Say for instance I'm using I/Q to close a PLL. In...

(how) does coherent sampling (used primarily in A/D testing?) differ from I/Q demodulation? Say for instance I'm using I/Q to close a PLL. In this case the 'sine wave' I'm locked to is being sampled at x4 the sine frequency. This would seem to satisfy the coherent sampling requirement that regarding the digitizer clock. next question - coherent sampling removes the need for windowing in FFT ap...

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web...

Hi Guys, in a recent thread, mention was made of a "Sampling" paper by Dan Lavry. At the following web site http://www.lavryengineering.com/pdfs/sample.pdf you can see Dan's 1997 paper: "Sampling, Oversampling, Imaging and Aliasing - a basic tutorial". I recommend caution if you decide to read that paper. In the second paragraph Dan wrote: *** Sampling theory *** ...

Hello, i feel confuse with sampling theorem. I know that if i respect sampling theorem : using sampling frequency above 2*the lowest...

Hello, i feel confuse with sampling theorem. I know that if i respect sampling theorem : using sampling frequency above 2*the lowest frequency inside the incoming signal then the signal can be reconstructed from the sequence of values (excuse my poor english). In other word no information is lost. But, i have to resolve another case. Some computation must be done on a signal at 100 Hz lik...

To all you DSP mavens out there. I am looking for some references to distortions introduced by sampling = errors, especially sampling close...

To all you DSP mavens out there. I am looking for some references to distortions introduced by sampling = errors, especially sampling close to the Nyquist criterion. In all the = literature I have seen, reconstruction of a waveform is guaranteed if the sampling = meets the Nyquist criterion, that is sample rate > 2F. =20 There are lots of cases where the Nyquist criterion doesn't seem

Hy, I recently read this good article of Richard Lyons but there is something i can not understand. Does someone could help me understand...

Hy, I recently read this good article of Richard Lyons but there is something i can not understand. Does someone could help me understand this difficulty ? It it says : "Each A/D converter operates at HALF the sampling rate of standard real-sampling" !!! How does it work ? It seems that using half the sampling rate of standard real-sampling, we obtain the same frequency definitio...

Hello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for...

Hello! I need to implement a low pass filter with a flexible cut-off frequency of 10-50Hz for signals sampled at 40kHz. I believe that for this approach a multirate and multistage filter is the best choose. So first the signal is down-sampled by a factor of approximately 100, then the signal is filtered with an IIR filter and finally the signal is up-sampling (may up-sampling is not required)...

I need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling...

I need to implement a particular filter that has 240 taps and it works/designed for a sampling rate of 125kHz. I want to change the sampling rate of the filter from 125kS/s to 250kS/s to match the the hardware I am using, which operates at 250kS/s if i run the filter as is (at 250kS/s), the spectrum of the filter is stretched by 2. Is there an easy way to modify the filter taps to change ...

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...

Hi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on...

Hi, it's well known that sampling a continuous function according to the sampling theorem requirements, you get all the information on the contunuous function just from the samples. now, if you have to get an accurate estimate of the integral of the continous function from samples satisfying the sampling theorem requirements, how can you get a good estimate of such an integral? I noticed that...

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...

Given I have an input sequence [ 1 0 1 0 1 0 1 0 ], the DFS is x[n] = 0.5 + 0.5*exp(j*pi*n). The sampling (Ts) is 100 secs. Am I correct in...

Given I have an input sequence [ 1 0 1 0 1 0 1 0 ], the DFS is x[n] = 0.5 + 0.5*exp(j*pi*n). The sampling (Ts) is 100 secs. Am I correct in saying that the spectral coefficient 0.5 at pi is equivalent to saying the DFS has a 0.5 peak every 200 secs? My reasoning is between adjacent x[n] samples a time Ts has passed which corresponds to 2*pi of the sampling period. Therefore, half, i.e....

I'd like to ask a couple very basic questions. Being hands-on rather than mathematically-focused I'm trying to visualize sampling in a project...

I'd like to ask a couple very basic questions. Being hands-on rather than mathematically-focused I'm trying to visualize sampling in a project I'm planning to start. 1. By sampling say a 3.1 kHz band-limited voice channel at the Nyquist rate am I guaranteed to capture *all* the information in the channel? Is nothing lost (ignoring ADC resolution)? More to the point: To know what the ...

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...

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

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...

Hi, I am new to signal processing, so please don't laugh if my questions appear silly to you. In my knowledge, to get accurate spectrum...

Hi, I am new to signal processing, so please don't laugh if my questions appear silly to you. In my knowledge, to get accurate spectrum result using FFT, the sampling period should be close to the integal times of the whole cycles of input signal period and the sampling points should be 2^n. For example, 1KHz square wave, the sampling time should be integal times of 1ms, i.e., 10ms. How...

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it....

Good Day, First my apology for not using right terms(not familiar) I have bought a c6711 from ebay. I need to know some thing about it. What is the maximum sampling rate I could achieve. I have used sound card in Linux for data acquisition(voice) maximum sampling speed with my laptop was 140000 samples/sec which means i could analyze frequencies up to 140000/2. now I want to use c671...

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in...

"Fourier Transform" transform the time domain data into frequency domain information on the premise that the sampling rate is constant or in some predictable manner. If the sampling clock jitter, it introduces noises into the frequency domain information. I would like to know, is there is formula whereby I can calculate the amount of noise introduced, based on how much the clock jitter ?