Let's say you have a perfect sine wave at frequency w. According to Nyquist, in order to be able to recover the sine wave, you need to have a...

Let's say you have a perfect sine wave at frequency w. According to Nyquist, in order to be able to recover the sine wave, you need to have a sampling rate of at least 2w. So if you decide to sample at 2w, you end up with 2 samples for each cycle of this sine wave. If you sample at the peaks and troughs (90 and 270 degrees) of the sine wave in time (or spatial) domain, you indeed preserve t...

Hi all, My project is Digiatal Sampling correction Interpolation/Decimation for application in flexible OFDM Modem. I'm implementing in Matlab...

Hi all, My project is Digiatal Sampling correction Interpolation/Decimation for application in flexible OFDM Modem. I'm implementing in Matlab the 5_taps optimized interpolator which will improve the performance for long data OFDM bursts. I used control parameters basepoint index (m(k)) and fractional interval (mu(k)) (reference to FM Gardners article "Interpolation In Digital Modems---PART1: ...

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM...

hi, Can someone explain how I can compensate for sampling clock frequency offset (SCO) using fractional-delay interpolation? In OFDM basebnad receiver design book they said that you can use fractional-delay interpolation filter to compensate SCO but of course they forgot to write about. Let's say I have a ramp signal (it will be easier for me to explain what my doubts are as it has amplitude...

Hi there, I have modelled a system that non-uniformly samples a bandpass signal. Unfortunately I am not allowed to transform to the...

Hi there, I have modelled a system that non-uniformly samples a bandpass signal. Unfortunately I am not allowed to transform to the equivalent bandpass. Now I have the problem that I should also model the samples, when bandpass filtered White Gaussian Noise is sampled. For uniform sampling, the variance of the Gaussian Variable of each tap equals N_0B. How can I evaluate the variance of t...

Hi All, If an analog signal is sampled such that sampling satisfies the Nyquist criteria, then is it possible to reconstruct the...

Hi All, If an analog signal is sampled such that sampling satisfies the Nyquist criteria, then is it possible to reconstruct the original signal back from the sampled signals perfectly? Is it really possible to get back the original signal 100% exactly as it was before? How do we reconstruct the analog signal given the sampled signal at the input side. Which interpolation function is the ...

Hi, I have successful using Mathlab created the Flat-topped PAM signal and magnitude spectrum. However, I am stuck with how to create the...

Hi, I have successful using Mathlab created the Flat-topped PAM signal and magnitude spectrum. However, I am stuck with how to create the natural sampling PAM signal and magnitude spectrum. Anyone who could guide me. Below are the details of the program and I have created the function program to link each other. Program 1(program1.m) %data fs=8000; %sample frequency ts=1.25e-4; %1/f...

Hi, I'm quite new to IQ-data and signal processing in general and am having some problems regarding which sample rate I actually must use when...

Hi, I'm quite new to IQ-data and signal processing in general and am having some problems regarding which sample rate I actually must use when I sample a signal and want to get IQ-data after signal processing on an FPGA. Lets start with an example signal that goes from 0-10 MHz. Nyquist criterion=> fs > 20 MHz Question 1: Must I sample in 20 MHz and then mix and decimate to get 10MHz

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I want to know what affect the sampling frequency of the audio will have on such a filter design. In other words, I can sample at 16kHz, filter the data and downsample to 8kHz (which is finally the sampling rate I want) OR I can sample at 8kHz and fi...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I want to know what affect the sampling frequency of the audio will have on such a filter design. In other words, I can sample at 16kHz, filter the data and downsample to 8kHz (which is finally the sampling rate I want) OR I can sample at 8kHz and fi...

Hi! I want to simulate a Ulatra Wideband Band system with 2GHz band and sampling frequency of 20GHz. The data rate is only 1kbps. This takes...

Hi! I want to simulate a Ulatra Wideband Band system with 2GHz band and sampling frequency of 20GHz. The data rate is only 1kbps. This takes huge amount of simualtion time, I wonder is there any way, I can reduce the simulation time by using some assumptions. I understand that in narrowband system, we can do some appropriate scaling in the sampling frequency and data rate and the results ...

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the spectrum of the...

on what basis the fft length is calculated related to the sampling frequency?I want to search for a maximum peak by obtaining the spectrum of the signal.I am working on demodulator in that my task is to acquire the carrier from the receive modulated signal.Modulation used is BPSK.It can be used for 300,600,1200,2400,10800,5400 bit rates.sampling frequency is 518400 for 10800 and 5400 bit rates,for...

Hi all, I am another guy getting stuck with the same old stuff! The relationship between point in a frequency response and the sampling...

Hi all, I am another guy getting stuck with the same old stuff! The relationship between point in a frequency response and the sampling frequency/time characteristics of the impulse response. I have created a custom frequency array, representing a frequency response from an algorithm, between 0 and 300Hz, with 0.2Hz resolution, ie 1500 points. I then need to create the impulse response from ...

Hi Friends, I am new to this. My questions is very basic. How do i decide that i need to use 1024 point FFT or 2048 point FFT or N point FFT...

Hi Friends, I am new to this. My questions is very basic. How do i decide that i need to use 1024 point FFT or 2048 point FFT or N point FFT for a system. I am working on a project, where my sampling frequency is 80mhz and i need to decide how many point FFT to be use? What are the parameters i need to take into considerations to decide the No. of points in FFT? Can anyone please help ...

Hi all, I made a quadrature tone decoder like this : I first sampling the input signal at 300 kS/s from an 8 bit ADC Then I mix this...

Hi all, I made a quadrature tone decoder like this : I first sampling the input signal at 300 kS/s from an 8 bit ADC Then I mix this signal with sine and cosine at the frequency I want to decode (22 kHz) Then I have a decimate filter to sampling at 60 kHz, then I can lowwpass filter at cutoof frequency of 500 Hz with an IIR filter to remove the other tone frequency then I compute the...

Suppose we have an ADC with three inputs (multiplexed). The ADC takes each input in turn. Suppose the frequency that the ADC scans ...

Suppose we have an ADC with three inputs (multiplexed). The ADC takes each input in turn. Suppose the frequency that the ADC scans bewteen inputs is fd. How should fd be related to the sampling rate per channel fs? 10 times bigger or 100 times bigger? and if the scan rate is too slow what effect does this have. Obviously the inputs cannot be matched in time because of the scan rate. Hard...

I want to change sampling rate of a narrow band signal by a fractional factor of 15/13. The original signal is -60dB or lower after f=0.08...

I want to change sampling rate of a narrow band signal by a fractional factor of 15/13. The original signal is -60dB or lower after f=0.08 (1 corresponds to half the sampling rate). I use matlab upfirdn function and design the anti-imaging/anti-aliasing filter using remez. The problem is that i get small droop in the passband no matter how sharp a filter i have got. I know this sounds a bit vague ...

I have developed a function which down samples an input signal.Say I have an input signal with a sampling rate of 512 samples/sec and would like...

I have developed a function which down samples an input signal.Say I have an input signal with a sampling rate of 512 samples/sec and would like to down sample it 128 samples/sec, then the down sampling factor is discrete and it is 4. In this case, I believe that the input signal will successfully be down sampled to 128 samples/sec without any loss of information.But, what if we have to down sampl...

Dear All, In page 8 of http://www.cs.tut.fi/kurssit/TLT-5806/Synch.pdf it is stated: All-digital symbol timing recovery scheme requires...

Dear All, In page 8 of http://www.cs.tut.fi/kurssit/TLT-5806/Synch.pdf it is stated: All-digital symbol timing recovery scheme requires higher-than-symbol-rate sampling. If this statement is correct what should be the minimum ratio? We have also a constraint on the max sampling rate of the ADC. It is 2.2x symbol rate. If we sample at 2.2x symbol rate should we also adjust the Matched filter...

I'm trying to perform Gibbs sampling. I've written the algorithm but I'm not sure how to calculate the jump/step size so that successful...

I'm trying to perform Gibbs sampling. I've written the algorithm but I'm not sure how to calculate the jump/step size so that successful iterations would be independent (Gibbs is MCMC). I have the data using a step of 1 and I've calculated the autocorrelation of the data. So I thought the jump size is the size when the autocorrelation crosses the x axis but this value seems to depend on...

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer...

Hi, everyone! I read "decimation by integer" and "interpolation by integer factors" in a DSP book. I'm just wondering why integer factor to downsampling... Sometimes, to resample signal by rational factor, people use conversion oversampling(integer factor) -> low-passfilter -> downsampling(also integer factor). (This means "Sampling rate conversion by non-integer factors") Howe