Problem in Bandpass filter, passband reappear after half sampling frequency

Started by in Matlab DSP19 years ago 3 replies

Hi all, I am beginner in digital filter design. Currently, I am designing a bandpass filter cut-off at 500hz and 1.8khz, with...

Hi all, I am beginner in digital filter design. Currently, I am designing a bandpass filter cut-off at 500hz and 1.8khz, with sampling frequency 6k Hz. I use butterworth filter with order 6 according to the specification. However, when I try to plot the signal, there is

Variable Rate Sampling

Started by in Matlab DSP18 years ago

Hi, I have to complete my DSP project "variable rate sampling (audio)" till the 10th of December,2003. I have been instructed that...

Hi, I have to complete my DSP project "variable rate sampling (audio)" till the 10th of December,2003. I have been instructed that I have to take the input directly through mic and process it in MATLAB. I am using MATLAB 6.5 release 13. I know the algorithm and commands for

Fsk code

Started by in Matlab DSP14 years ago

Hi,I would like to ask about my Fsk code.Why when i change my ts value to 0.05 or 0.005,i'm not able to get the correct Fsk signal?Anybody can...

Hi,I would like to ask about my Fsk code.Why when i change my ts value to 0.05 or 0.005,i'm not able to get the correct Fsk signal?Anybody can help me?Below is what my code looks like.. echo on t0=0.6; %signal duration ts=0.0005; %sampling interval fc=200; %carrier frequency kf=50; %modulation index fs=1/ts; %sampling frequency t=(0:ts:t0); %tim...

recursive frequency sampling filter design

Started by in Matlab DSP19 years ago 1 reply

Dear all, I am working on the recursive frequency sampling filter design. My confusion is, I read the theory saying that as...

Dear all, I am working on the recursive frequency sampling filter design. My confusion is, I read the theory saying that as it contains the poles.and also exactly pole zero cancellation occurs to give the stability. in case of fir filter using remez or any,

Peakvue simulation in MATLAB

Started by in Matlab DSP15 years ago

hi all, actually i'm still newbie here compare to all of you. i have a question about peakvue.is there anyone have ever try to simulate for...

hi all, actually i'm still newbie here compare to all of you. i have a question about peakvue.is there anyone have ever try to simulate for showing peakvue measurement in MATLAB? i think the process can be describe as: 1. sampling (we choose Fmax as a limit to show in freq domain and the sampling freq will be automatically determined 2.56*Fmax) 2. pass the signal through high pass fil...

How to record sound in Matlab?

Started by in Matlab DSP20 years ago 1 reply

Hi everybody, I'm working on a project on speech recog. How do you record sound in matlab at a sampling freq of 8192Hz. Currently...

Hi everybody, I'm working on a project on speech recog. How do you record sound in matlab at a sampling freq of 8192Hz. Currently i'm using windows sound recorder to record the sound at a sampling freq of 22050 Hz which is too much for me. format conversions leads to inaccurate

Hilbert transform

Started by in Matlab DSP16 years ago

All Background: I have working with ultrasound signals (300 Khz) from a solid state sensor to a target at 3". The received signals from...

All Background: I have working with ultrasound signals (300 Khz) from a solid state sensor to a target at 3". The received signals from the sensor (300 Khz) are digitized through a sampling card and imported into MATLAB, to preserve the time information as precisely as possible, sampling frequency is high Fs = 50Mhz. To generate a envelope on the 300Khz received

random row sampling

Started by in Matlab DSP14 years ago 2 replies

Hi all, Suppose I have a matrix X as follows, 2 3 4 1 8 7 6 2 1 2 5 7 4 7 2 ...

Hi all, Suppose I have a matrix X as follows, 2 3 4 1 8 7 6 2 1 2 5 7 4 7 2 3 Then how should I write a code to sample two row ramdomly (called Y) from that matrix. At last, I must have two matrix; X matrix must become 2x4 (with remaining row after sampling) and Y matrix (2x4). Please help me as soon as possible. Thanks i...

how to convert recorded speech sampling?

Started by in Matlab DSP18 years ago 3 replies

i'm new here, i am doing my university project on speech recognition, if somebody could tell me how to convert recorded speech...

i'm new here, i am doing my university project on speech recognition, if somebody could tell me how to convert recorded speech sampling? i've already try using resample code from matlab, but the result was that the sound of speech file dosen't sound very nice. the code that i use was

recursive frequency sampling filter design

Started by in Matlab DSP19 years ago

Dear all, my problem straight away, i am doing recursive frequency sampling filter design, in matlab i design the...

Dear all, my problem straight away, i am doing recursive frequency sampling filter design, in matlab i design the filter coefficients, i gave the format long at the top and received the 15 digits for representation after decimal point. does this give

Minimum lag high pass filter for very low frequencies

Started by in Matlab DSP9 years ago

Hello, I was hoping I could get some help on designing a high pass filter for very, very low frequency signals (relative to the sampling...

Hello, I was hoping I could get some help on designing a high pass filter for very, very low frequency signals (relative to the sampling rate). The frequencies I'm looking at are: 20Hz sampling, I want to filter off everything below 1/30Hz (or 1/15 if that's the best I can do). My application is very lag and phase shift sensitive. I have to double integrate this filtered signal, and the bett...

am modulation - different sampling frequency

Started by in Matlab DSP14 years ago 1 reply

Hello everybody, I am writing a simple am modulator (actually ssb, but it does not really matters :) ). This script shall take as input an audio...

Hello everybody, I am writing a simple am modulator (actually ssb, but it does not really matters :) ). This script shall take as input an audio signal and give as output the am modulated signal. The problem I am facing is the following. The input audio signal as, for instance, a sampling frequency of 8kHz. In the am modulator script I have much higher frequency rates: the carrier has a frequ...

converting variable sampling rate to fixed rate

Started by in Matlab DSP14 years ago 1 reply

I am working with a signal that was sampled at a variable rate and consists of an array of values and a separate array of sample times. The...

I am working with a signal that was sampled at a variable rate and consists of an array of values and a separate array of sample times. The sample intervals are random and are not multiples of any common value. I need to convert this signal to a single array with a known, fixed sampling rate. Does anyone have suggestions as to how I would go about this? Thanks, Chris

Started by in Matlab DSP15 years ago

hai friends i have problem in modelling a rayleigh fading channel. in matlab one command for rayleigh channels ,in that the...

hai friends i have problem in modelling a rayleigh fading channel. in matlab one command for rayleigh channels ,in that the parameters sampling rate , dopplershift and delay spread. i wnat a impluse reponse of the chhanel for some sampling rate and delay spread. how can i proceed? one more is i have to model the 3 or 4 tap channel. given data was just delay spread. how can choose delay v...

How to plot freq response for a combined filter

Started by in Matlab DSP18 years ago

Hi, Can anybody help me plot a combined frequency response of 2 filters in Hz? Hk=Hk1.*Hk2; The sampling freq for Hk1 is 96Khz...

Hi, Can anybody help me plot a combined frequency response of 2 filters in Hz? Hk=Hk1.*Hk2; The sampling freq for Hk1 is 96Khz and the sampling freq for Hk2 is 24Khz. figure(1) plot(96e3*w/(2*pi),20*log10(abs(Hk2)),'y'); hold on figure(1) plot(24e3

problem with IFFT and sampling time

Started by in Matlab DSP16 years ago

Hi everybody, My problem is as follows: I get data from a fft analyser after numerous averaging in order to remove the noise. Now in...

Hi everybody, My problem is as follows: I get data from a fft analyser after numerous averaging in order to remove the noise. Now in order to use it i have to take the IFFT of this data in Matlab. But fft analyser(any) while taking fft take sampling frequency as 2.56 times the maximum frequency specified. But when IFFT is taken in matlab it will just do

PSD....

Started by in Matlab DSP14 years ago

Can someone send me a sample code to do PSD of a signal in matlab, the sampling frequency is 250hz. regards, Srinivas

Can someone send me a sample code to do PSD of a signal in matlab, the sampling frequency is 250hz. regards, Srinivas

Re: Plotting a signal

Started by in Matlab DSP17 years ago

Hi, May be you can try this fs = 500e3; % sampling freq N = length(y); TimeWindowLength = N/fs; % Time axis range deltaT =...

Hi, May be you can try this fs = 500e3; % sampling freq N = length(y); TimeWindowLength = N/fs; % Time axis range deltaT = 1/fs;%distance between samples x = 0:deltaT:deltaT*N; plot(x,y); I know this can be done with less steps... but this will give u an understanding of the problem... You can extend this furthe

interpolation function

Started by in Matlab DSP18 years ago

HI, I?m having a problem with some things I'm doing in matlab. I have a function whose sample frequency is not constant, I have to ...

HI, I?m having a problem with some things I'm doing in matlab. I have a function whose sample frequency is not constant, I have to interpolate it so as to obtain a constant sampling freq. I'm trying to use the interp1, but I'm not having good results. Is there anybody who can help me

Quantization and Stationarity

Started by in Matlab DSP20 years ago

hi everybody i would appreciate if someone could suggest any solution to the following problems: 1)Find the number of bits...

hi everybody i would appreciate if someone could suggest any solution to the following problems: 1)Find the number of bits needed to achieve a signal-to-quantization noise ratio of at least 80dB for a sampling device calibrated so that the sampled intensitie