Hello, I have question regarding sampling. I've converted a lead/lag from the s domain to the z domain using the bilinear method. The...

Hello, I have question regarding sampling. I've converted a lead/lag from the s domain to the z domain using the bilinear method. The processing time 100msec and the input update time of the A/D is 120msec, asynchronous to the processing time. Should I set the processing time to 120msec or 125msec to atleast ensure that an update will occur at the start of the lead/lag processing? T...

It is stated that in case of narrow transition band filters, multirate filtering provides us with a computational efficiency as compare...

It is stated that in case of narrow transition band filters, multirate filtering provides us with a computational efficiency as compare to standard time invariant filters. The idea is to reduce the sampling rate (less number of samples) and to use simple, low order filters (less number of operations per input sample), instead of using a single high order filter operates at fixed high sampling rate...

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---PART...

HI all, I am trying to explore the effect of using sine wave clock from sampling instead of conventional square wave. I understand that the...

HI all, I am trying to explore the effect of using sine wave clock from sampling instead of conventional square wave. I understand that the major problem will be due to slow rise time of sine wave as compared to square wave but at high frequency sine wave still has sufficiently good rise time. Or may be we can make the slew rate of both sine and square wave same. All this apart, i have follo...

I have a decimation filter which decimates by 100. Input sampling frequency fclk1 = 1.6MHz, Output Sampling freq flck2 = 16kHz. I am...

I have a decimation filter which decimates by 100. Input sampling frequency fclk1 = 1.6MHz, Output Sampling freq flck2 = 16kHz. I am trying to plot the input and output frequency responses. For the input, I use fft function in matlab, and get the magnitudes of my frequency response. To generate the plot X-axis points (i.e., frequencies) I do the following freq = (flck1/fftPtsIn)*(...

If I have sampled a signal at say fs=1MHz sampling rate and I capture N=100 samples of that signal, my resolution bandwidth in frequency domain...

If I have sampled a signal at say fs=1MHz sampling rate and I capture N=100 samples of that signal, my resolution bandwidth in frequency domain will be fs/N = 10kHz. 100 samples correspond to 100us of capture. If sampling frequency is doubled and if the capture duration is also doubled, I would expect the resolution bandwidth also to increase by a factor of x. But it is not the case. Reso...

Hi all, Suppose x(t) has bandwith bandlimited in [-B, B]... so the lowest sampling rate for x(t) is Fs=2B... Does this matter when x(t) is...

Hi all, Suppose x(t) has bandwith bandlimited in [-B, B]... so the lowest sampling rate for x(t) is Fs=2B... Does this matter when x(t) is real or complex-valued? Moreover, for (x(t))^2, the bandwidth is [-2B, 2B], the lowest Fs=4B no matter when x(t) is real or complex-valued. Am I right? More interestingly, for (x(t))^3, the bandwidth is [-3B, 3B], but we can still use Fs=2B(th...

Hi: I=2E Audio vs. Video Digitized (mono) audio has a single sample per each sampling interval. In the case of digital video, we could...

Hi: I=2E Audio vs. Video Digitized (mono) audio has a single sample per each sampling interval. In the case of digital video, we could treat each individual sample point location in the sampling grid (each pixel position in a frame) the same way as if it was a sample from an individual (mono) audio signal that continues on the same position in the next frame. For example, a 640=D7480...

How to design Decimator with the extra filter A(z) at the output and Interpolator with the extra filter A(z) at the input. .. x(n) is the...

How to design Decimator with the extra filter A(z) at the output and Interpolator with the extra filter A(z) at the input. .. x(n) is the input Fs = 24 in the input sampling rate M= 6 is the number of subfilter In case decimator A(z) is operated at output sample rate . as Fs/M ..where M is 6 .Fs is the input sampling rate. In case Interpolator A(z) is placed at the input and operates at in...

Hi everyone For a mini project I am doing, I want to generate Doppler coefficients for Jakes model. But I want to do this with varying sampling...

Hi everyone For a mini project I am doing, I want to generate Doppler coefficients for Jakes model. But I want to do this with varying sampling rates, i.e. generate at twice the maximum Doppler frequency in 1st iteration and then at 5 times the maximum Doppler in the 2nd iteration etc. I read the book "Mobile fading Channels by Matthuas Patzold", but in the methods he has given I could not fin...

Hi, I am trying to process audio came from microphone using FFT. But audio driver gives only 512 bytes in a buffer. Sampling rate...

Hi, I am trying to process audio came from microphone using FFT. But audio driver gives only 512 bytes in a buffer. Sampling rate is 44Khz. Whether to do FFT complete sample for 1 sec is required ? Or I can process these 512 Bytes buffer one by one ? If yes any examples on net ? I already have a Function which takes buffer and sampling rate and does FFT. (i.e from Numerical Recipies usin...

I would like to use an audio codec based on WAVE PCM. It should be a little different though. The bit-resolution should be set to...

I would like to use an audio codec based on WAVE PCM. It should be a little different though. The bit-resolution should be set to equal 1/(sampling rate X # of channels). The bit-rate should be set to equal 1 bit per second. I would like to use this codec to transport audio files though the internet via email. I am looking for frequency response. In digital audio the sampling rate must be ...

Hi, I need to extract a signal from an input signal contaminated by noise and interferences. The input signal is sampled at 100MHz. The...

Hi, I need to extract a signal from an input signal contaminated by noise and interferences. The input signal is sampled at 100MHz. The signal I want resides from 0Hz to 500KHz only, and noise and interferers are everywhere. In order to extract the narrow band signal in 100MHz sampling rate, I am thinking of LPF and then downlsampling but I don't know how to decide the number of stages and ...

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; ...

clear all; Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 seconds. t=0:1/Fs:T; f0 = 50; % Initial frequency is 50hz f1 = 100; % f(t) changes from 50hz to 100Hz in 2 seconds. freq = f0 + (f1-f0)/T*t; y = sin(2*pi*freq.*t); % Chirp signal y(t) Fs = 1000; % Sampling frequency T = 2; % Signal time duration is 2 sec...

I have a basic question here regarding OFDM digital front end. if i have a very efficient clock of say 10ppm, can i downsample the signal to the...

I have a basic question here regarding OFDM digital front end. if i have a very efficient clock of say 10ppm, can i downsample the signal to the sampling frequency of transmitter. i.e. the sampling frequency at which i will get 64 samples for a 64 sample FFT block? for example, the FFT output at the transmitter is at 500 kHz. Now can i directly down sample this signal to 500 kHz baseband sign...

Hi, I am trying to reconstruct bi-phase digital data from bi-phase analog signal. I am sampling analog signal at 4 times of actual data rate....

Hi, I am trying to reconstruct bi-phase digital data from bi-phase analog signal. I am sampling analog signal at 4 times of actual data rate. I am getting ADC output for each sampling instant. But, I have no idea about the bi-phase reconstruction logic from sampled ADC output. Any document document or suggestion would help me. Thanks in advance Rahul ______________________...

any one please can provide me with some information about how to upsampling and downsampling any function such as x (n) by 2 factor

any one please can provide me with some information about how to upsampling and downsampling any function such as x (n) by 2 factor

can anyone clarify which of the following is the right method for re-sampling (sample rate conversion) method 1 - up-sample by adding...

can anyone clarify which of the following is the right method for re-sampling (sample rate conversion) method 1 - up-sample by adding additional samples (usually 0) to the input samples - convolve using a LPF method 2 - up-sample by adding additional samples to the input samples - pick up only selected samples (e.g. 1 in 10; 1 in 20 etc.) I am unable to appreciate about method 1 since...

Fred, Blame Google Groups. I don't grok it. Jerry

Fred, Blame Google Groups. I don't grok it. Jerry

Is there a way to take a 1d wavelet trans to get 2 bands (lpf and hpf) over a fixed BW (say 4KHz) if the sampling freq is greater (eg 20KHz)

Is there a way to take a 1d wavelet trans to get 2 bands (lpf and hpf) over a fixed BW (say 4KHz) if the sampling freq is greater (eg 20KHz)