Hi, all. I have a problem to compensate "sampling phase error" In spectrum analyzer, I get the sampling data from analog signal to...

Hi, all. I have a problem to compensate "sampling phase error" In spectrum analyzer, I get the sampling data from analog signal to OFDM signal. but I knew that there is sampling phase error. I mean, sampled data can be expressed as y[0], y[1], y[2].....for time domain signal. but, it is not exact point. In some processing sampled data with C language, there were some phase error....

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at...

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at 1048 Hz in which i want to suppress everything above 50 Hz so i make a low-pass FIR filter (i know a0, a1,.... filter coefficient). But i want to keep the same filter behaviour with the same signal but sampling this time at 10008 Hz for example. ...

Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320...

Hi all, I would like to know if there is an easy method to do the following: I have a signal of, say, 10ms, sampled at 32kHz (320 samples). I would like to convert this to a signal sampled with a varying sampling rate, for example, starting at 16 kHz, going down to 4 kHz at the end. This would involve a time-variant anti-alias-filter and non-uniform sub-sampling. Is there a standard...

how to generate a pulse train for sampling a signal?

how to generate a pulse train for sampling a signal?

Dear All, I have one problem in real time signal processing. My signal frequency is 4000Hz and the sampling frequency is 8000Hz. Actually it...

Dear All, I have one problem in real time signal processing. My signal frequency is 4000Hz and the sampling frequency is 8000Hz. Actually it should be more than 8KHz. Becasue of that i am loosing the zero resting points of the sampled signal. With out increasing the sampling frequency , is there any remedy for avoiding the information loss ? Thanking you,

Hello, I have two filters, one designed for a sampling frequency of 16kHz and the other for a sampling frequency of 8kHz. The type and kind...

Hello, I have two filters, one designed for a sampling frequency of 16kHz and the other for a sampling frequency of 8kHz. The type and kind of filters (FIR/IIR, HP/LP/Notch) is irrelevant. I need to plot the overall transfer function of this cascaded pair. What changes do I have to make (in the 8kHz filter, I am assuming), if any at all, in order to cascade the two filters? Pointers to ...

Hi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the...

Hi all, Suppose a signal is known a prior. Time domain as well frequency domain characteristic is pretty defined. Say fs is the sampling frequency as per Nyquist criteria. Number of samples in 1 sec is N. Let us think two cases 1. Uniform sampling i.e sample interval is constant, I guess this is the usual case in most of the applications. 2. We keep N same in I sec but a variabl...

Hello all, I am reading a matlab example from Simon Haykin's introduction DSP book. In the example, a discrete time signal is describe...

Hello all, I am reading a matlab example from Simon Haykin's introduction DSP book. In the example, a discrete time signal is describe as x[n] = e^-(n/15) sin(2*pi*n/13 + pi/8) for 0

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)...

A Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and...

A Sound Mathematical Basis For Sampling - Lesson 3 -------------------------------------------------- Good Morning, once again, Boys and Girls! I'm sorry that I got called away yesterday; SWMBO, indeed, MBO! Today I'll derive for you the mathematics of sampling, based on an analysis of the circuits that we actually use, rather than on some dubious mathematics (I will show later why i...

Hi, I am using 444 kHz sampling frequency (fs) to sample two different carrier frequencies: f1=108 kHz and f2=111 kHz. When I look at the...

Hi, I am using 444 kHz sampling frequency (fs) to sample two different carrier frequencies: f1=108 kHz and f2=111 kHz. When I look at the sampled f2, the waveform seems to be pretty accurately represented. However, for f1 this is not the case. Even thought the frequency of f1 is represented accurately, the amplitudes seem to oscillate. My interpretation is that because fs is not an intege...

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...

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...

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...

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

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...

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...

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.

Hi All, I am new to DSP and I have a very basic question (Figure 13 of Rich Lyons tutorial Quadrature sampling complex but not complicated) -...

Hi All, I am new to DSP and I have a very basic question (Figure 13 of Rich Lyons tutorial Quadrature sampling complex but not complicated) - When we have a signal with center frequency f_0, why does sampling it with frequency f_s and digitizing the signal result in the spectrum of the signal being now centered at f_s and - f_s? I understand that multiplying two signals f_1 and f_2 resu...

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