SUI channel resampling

Started by Quoc Lai in Matlab DSP10 years ago 1 reply

Hello, Could anyone let me know how I would resampling the generated SUI channel to obtain a desire sampling rate? Thank you, Quoc ...

Hello, Could anyone let me know how I would resampling the generated SUI channel to obtain a desire sampling rate? Thank you, Quoc


OFDM frequency spectrum

Started by Hussain in Matlab DSP12 years ago 1 reply

Hello guys, I don't know a lot about matlab simulink, so I was wondering if someone can help me so I can figure out the spectrum of an ofdm. I...

Hello guys, I don't know a lot about matlab simulink, so I was wondering if someone can help me so I can figure out the spectrum of an ofdm. I want the complex data out from the transmitter to be analogized separately- I mean the real and imaginary signals each one alone- then digitized by a sampling rate that is twice the value of the digital transmitted signal. Thanks in advance, Huss...


Uniform sampling with min and max

Started by Anonymous in Matlab DSP20 years ago 2 replies

Hi Given that I have a matrix, say A = [1 2 3; 4 5 6], how do I draw new samples over the range of A columns using uniform ...

Hi Given that I have a matrix, say A = [1 2 3; 4 5 6], how do I draw new samples over the range of A columns using uniform distribution? For example, min(:,1) = 1 and max(:,1) = 4 Therefore my new sample can have data points ranging from 1 to 4 in the first


Multi-rate Fir Filter Design

Started by c r in Matlab DSP19 years ago 1 reply

I am designing an audio equalizer in Matlab, and I want to do it with a multirate setup, but I do not want to sample the input wave...

I am designing an audio equalizer in Matlab, and I want to do it with a multirate setup, but I do not want to sample the input wave more than once, so if I have a sampling of say 50khz, I was wondering what the best way to decimate the signal would be in order to make it work.  I plan on implementing th


Choosing ripples

Started by Anonymous in Matlab DSP21 years ago 4 replies

Hi firiends Please excuse me if I have posted this query on irrelevant discussion group. I have a basic question on Lowpass...

Hi firiends Please excuse me if I have posted this query on irrelevant discussion group. I have a basic question on Lowpass Filter Design. I am talking in the context of Upsampling and Down sampling. What is the criteria to choose the ripples in passband and stop band?


cut off frequency

Started by vani chezhiyan in Matlab DSP18 years ago 1 reply

hi all, i'm doing a project in preprocesing biosignals. if i'm given with a signal,how will i find out the cut off...

hi all, i'm doing a project in preprocesing biosignals. if i'm given with a signal,how will i find out the cut off frequency? is it possible only by looking into the power spectrum.is there any formula(given only the sampling frequency)? kindly help.  


RE: [speechcoding] DTMF Detection

Started by Hocquet Franck in Matlab DSP20 years ago

Hi Shehryar, The choice of N is mainly driven by the frequency resolution needed, which sets a lower boundary. N also is...

Hi Shehryar, The choice of N is mainly driven by the frequency resolution needed, which sets a lower boundary. N also is chosen so that (k/N)fs most accurately coincides with the actual DTMF frequencies assuming k are integer values and fs is a sampling frequency of 8 ks


comb filter and distortion

Started by egan_nc in Matlab DSP18 years ago

Hi, I am designing a 3 stages decimation filter for a second order 1 bit delta sigma modulator. The sampling rate of DSM is...

Hi, I am designing a 3 stages decimation filter for a second order 1 bit delta sigma modulator. The sampling rate of DSM is 1.536Mhz. The bandwidth for the input signal is 4Khz. For the first stage I used matlab's function cicdecimate with the following parameters:


HELP..ADCs in MATLAB

Started by engi...@yahoo.com in Matlab DSP15 years ago

Hello guys, I'm a new member in your group and I beleive that I can benefit from you a lot. I'm new to MATLAB and I need some help in...

Hello guys, I'm a new member in your group and I beleive that I can benefit from you a lot. I'm new to MATLAB and I need some help in implementing a function in MATLAB to do the job of the Successive Approximation ADC [saadc(vi, nbits, fs) where vi is the input analog voltage (between 0 and 5), nbits is the number of bits for this ADC (between 3-24), and fs is the sampling frequency]. Wa...


DFT spectral analysis

Started by sale...@gmail.com in Matlab DSP14 years ago 4 replies

Hi, I'm having trouble with a homework question. Suppose I'm given a signal x[n], which is obtained by sampling at a frequency of 2kHZ.Using...

Hi, I'm having trouble with a homework question. Suppose I'm given a signal x[n], which is obtained by sampling at a frequency of 2kHZ.Using matlab I plotted its spectrum (vs. frequency) and got the following: http://img141.imageshack.us/my.php?image=dft1cc2.jpg The question is (quote): "Can you retrieve any information about the distribution of frequencies as a function of time?" Wha...


dynamic time warping

Started by Lal Prasad in Matlab DSP18 years ago

hi, i'm doing aproject on voice recognition.java module is used for sound capturing and sampling and all are done by matlab.let...

hi, i'm doing aproject on voice recognition.java module is used for sound capturing and sampling and all are done by matlab.let me know how the sound written on the disk is taken to matlab as input(both matlaba nd java are running simultaniously).also can any of u contribute s


framing audio file

Started by smruts14 in Matlab DSP18 years ago

i have read a .wav file using waveread. it is a 16-bit dual channel with sampling frequency of 22.1KHz. i want to frame it into...

i have read a .wav file using waveread. it is a 16-bit dual channel with sampling frequency of 22.1KHz. i want to frame it into frames of 90ms duration.i have already put it in 1-d array. can anyone help me with the framing code....i tried using matlab help...but couldnt find anythin


Down converting Equation Please

Started by ahmad chaudhary in Matlab DSP14 years ago

Hi every one, First my apology for not using/knowing matlab termnalogy. One question how could I down convert in software. ...

Hi every one, First my apology for not using/knowing matlab termnalogy. One question how could I down convert in software. Suppose I have a data from real world @Sampling rate 100,000 per/sec. I know It has ultra sonic voice in it. How could I convert it to 20000 per/sec. Do I just drop every other signal? making it 50000? Can some one writhe me an equation? Sin...


step and impulse response help

Started by emkatsogridakis in Matlab DSP13 years ago 2 replies

Hi everyone. I m new to matlab and digital signal processing. So i need your help! As an excercise i was given the following signals alpha =...

Hi everyone. I m new to matlab and digital signal processing. So i need your help! As an excercise i was given the following signals alpha = .1; fs = 1; %sampling frequency [Hz] t = (1:256)./fs; %time array [s] x = exp(-alpha.*t); %input array x=e^(-at) y = ones(1,256); %initially set all values to be equal to 1 y(1:20) = zeros(1,20); %then set the first group of samples to zero I ...


16 QAM in Matlab

Started by kekl...@gmail.com in Matlab DSP10 years ago

Hi, I have to drew constellation diagram for 16 QAM modulation but I dont know how. Sampling frequency is 32khz and frequency of a carrier is...

Hi, I have to drew constellation diagram for 16 QAM modulation but I dont know how. Sampling frequency is 32khz and frequency of a carrier is 8khz. If you need something more ask me. Here is my code in CCS if that could help you: #include "tonecfg.h" #include "math.h" #include "dsk6713.h" #include "dsk6713_aic23.h" #define pi 3.14159 DSK6713_AIC23_Config config = { 0x0017, //...


90 degree phase shifter for 60Hz

Started by mehm...@gmail.com in Matlab DSP11 years ago

Hi, I?m trying to implement hilbert filter with a narrow ripple and my limited processing power. TW = 35; % Transition Width 35...

Hi, I?m trying to implement hilbert filter with a narrow ripple and my limited processing power. TW = 35; % Transition Width 35 Hz Apass = 0.02; % Passband Ripple (dB) -> variation 0.1% Fs = 1024; % Sampling Frequency h = fdesign.hilbert('TW,Ap', TW, Apass, Fs); Hd = design(h, 'equiripple'); Hhilb = FilterStructure: 'Direct-Form FIR' Arithmetic:


creating user defined blocks

Started by krishna_31985 in Matlab DSP16 years ago 4 replies

hi friends, i am doing my project on transmultiplexers.i want to do it in simulink.i am trying to create blocks for each one like...

hi friends, i am doing my project on transmultiplexers.i want to do it in simulink.i am trying to create blocks for each one like decimator,interpolator............. but i always get an error .the following is the interpolation program.can anyone tell me how to create a block for this in simulink clf; dt=0.00005; t=-0:dt:N-1; N=input('input length:'); L=input('up-sampling fa


Non-Uniform Sampling

Started by simha j in Matlab DSP21 years ago 1 reply

Hi all, My Question is: Is FFT applicable to a non-uniformly sampled signal?. I heard from one of my friends that we can...

Hi all, My Question is: Is FFT applicable to a non-uniformly sampled signal?. I heard from one of my friends that we can apply DFT to a N-US signal and not FFT. Is this correct? And also how is the nyquist rate defined for a non uniformly sampled sign


Ways to discretize continuous systems

Started by Raoul in Matlab DSP18 years ago

Hi to all, matlab people I'm searching for a function or a way to discretize continuous systems with Euler's method, that is:...

Hi to all, matlab people I'm searching for a function or a way to discretize continuous systems with Euler's method, that is: the constinuous variable is substituted with (1 - z^-1 ) / Ts ,where Ts is the sampling period and z^-1 is the discrete variable corrisponding to one s


zero padding question

Started by phil...@nrc.ac.uk in Matlab DSP14 years ago

I have a question that has been bugging me fo rsome time and I am hoping someone here can help me get to the bottom of it. It is a zero padding &...

I have a question that has been bugging me fo rsome time and I am hoping someone here can help me get to the bottom of it. It is a zero padding & FFT issue. Here goes; If I take 2 random signals for deconvolution (say 1000 samples and 1500 samples long with same sampling time). If I zero pad both out to 4096, take FFTs, devide one by another, then take iFFT, I get a graph of one signal deconvo...