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## Filter Impulse Response |

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impulse response , impulse response matlab , channel impulse response , matlab impulse response

83 Threads found on edaboard.com: **Filter Impulse Response**

I have a signal cos(w0 * t). It is sampled at frequency f_s. The samples are applied to a moving average **filter** of length L. I want to find the amplitude of the output.
So far I understand the following. Please do correct me if I am wrong; I am still learning the basics.
DTFT of **impulse** **response** of the moving average (...)

Digital Signal Processing :: 03-18-2017 05:48 :: Debdut :: Replies: **0** :: Views: **1115**

Hi i have attached an image of a simple low-pass **filter**. To the left of the image the **impulse** **response** is something decreasing in time, while the **impulse** **response** in the right is in the sinc function. I can understand the sinc function since the input **impulse** is a rectangular waveform which (...)

Analog Circuit Design :: 08-08-2016 20:18 :: preethi19 :: Replies: **1** :: Views: **2259**

I try to simulate a noisy signal and **filter** it. the signal mix some low frequency signals and some random noise. my goal is to get 14.8Hz signal. my band-pass bandwidth is 14.7Hz to 14.9Hz.
function **filter**ingTest
Hd = Kaiser**filter**;
Fs = 4000; % Sampling frequency
T = 1/Fs; % Sample time
L

Digital Signal Processing :: 04-30-2016 21:08 :: fences :: Replies: **5** :: Views: **1329**

Build FIR **filter** model using Verilog-A.
You can find many examples in "rfLib" of Cadence dfII.

Software Problems, Hints and Reviews :: 04-21-2016 04:02 :: pancho_hideboo :: Replies: **1** :: Views: **791**

A Linear Time-Invariant system with **impulse** **response** h1h1 is an ideal lowpass **filter** with cutoff frequency ωc=π/2ωc=π/2. The frequency **response** of the system is H1(ejω)H1(ejω). Suppose a new LTI system with **impulse** **response** h2h2 is obtained from h1h1 by (...)

Digital Signal Processing :: 02-28-2016 22:51 :: zer0square :: Replies: **2** :: Views: **937**

what is difference between **impulse** **response** and coefficient of digital **filter** in FIR.

Digital Signal Processing :: 10-13-2015 19:06 :: kushal nandanwar :: Replies: **0** :: Views: **907**

The question title involves a contradiction in terms, because a CIC decimator is a **filter** with finite **impulse** **response** as well.
So more exactly you are asking about using different FIR characteristics than CIC as a decimation **filter**. The general answer is quite simple, if you want a different frequency characteristic than (...)

Digital Signal Processing :: 05-31-2015 08:50 :: FvM :: Replies: **1** :: Views: **1435**

Hello.
I am unable to understand what you want to ask. I guess you are asking on developing a finite **impulse** **response** **filter**.. what you need is get the **filter** coefficients.. if the coefficient is in time domain, you just need to convolve with the signal you need to **filter**..

Digital Signal Processing :: 05-22-2014 20:40 :: u7karsh :: Replies: **1** :: Views: **807**

Hi friends,
I want to design a adaptive digital FIR **filter** using verilog.
I studied the base paper. But i am not clear with the multiplication of **impulse** **response** with the input sequence.
And i also dont get about the difference of Normal **filter**s and adaptive **filter**s.
Pls explain me briefly with (...)

PLD, SPLD, GAL, CPLD, FPGA Design :: 02-21-2014 03:59 :: Muthuraja.M :: Replies: **0** :: Views: **1092**

hello dear, i generated a target **impulse** **response** of fir low pass **filter** and then interpolated it , now i want to generate ternary tabs or coefficient using second order sigma delta modulation, please tell me how to generate it, if any one have code or idea

Digital Signal Processing :: 02-06-2014 05:26 :: chander kumar :: Replies: **0** :: Views: **643**

hello all , I need to implement CIC decimator, here is the matlab code using fvtool
r=64;
m=1;
n=3;
iwl=16;
owl=12;
IFL = 0; % Input fraction length
CIC = mfilt.cicdecim(r,m,n);
CIC.InputFracLength = IFL;
f_in =5120;
h=fvtool(CIC,'FS',f_in);
I checked the **impulse** **response** using the fvtool and it gives me a complete different respo

Digital Signal Processing :: 07-19-2013 12:02 :: ali kotb :: Replies: **0** :: Views: **940**

hello all ,
I have a problem when designing CIC **filter** on matlab
here is the code :
r=64;
m=1;
n=3;
IFL = 0; % Input fraction length
CIC = mfilt.cicdecim(r,m,n);
CIC.InputFracLength = IFL;
I use fvtool to check the impusle **response** and I got figure 1 attached ,
9330093299
I also use

ASIC Design Methodologies and Tools (Digital) :: 07-06-2013 22:44 :: ali kotb :: Replies: **0** :: Views: **928**

I am currently designing a Manchester matched **filter** design in Matlab. My task requires me to create and plot the **impulse** **response** h(t) of the optimum receiver **filter** matched to detect symbol '1'.
I have created a matched **filter** by using the following code:
h=fliplr(p);
where p is my manchester (...)

Digital Signal Processing :: 05-19-2013 01:45 :: hamonrye :: Replies: **0** :: Views: **2200**

Can annyone explain to me what is the **impulse** **response** for this half band **filter** mean? What is the purpose to simulate this **impulse** respone?
89782

Analog Integrated Circuit (IC) Design, Layout and Fabrication :: 04-22-2013 02:47 :: nhkch :: Replies: **0** :: Views: **579**

Guys,
I have created a **filter** from FDAtool and want to test it on Simulink but I didn't see any **response** on the Scope
I used discrete **impulse** as input What can I do to see the output ?
Thanks
I get this warning :
Warning: The model 'FIR**filter**Test' does not have continuous states, hence using the solver (...)

Digital Signal Processing :: 03-04-2013 12:17 :: bianchi77 :: Replies: **2** :: Views: **1329**

Hi, I'm trying to implement the overlap save method in matlab in order to clear up noise from a wav file. The function accepts the following fields:
x = long sequence to be **filter**ed (from wav file)
h = **impulse** **response** of **filter** (loaded from a different file)
N = Block length used in the algorithm ( i.e. size of the (...)

Digital Signal Processing :: 11-06-2012 21:58 :: Richie836 :: Replies: **0** :: Views: **3955**

Linear phase is in fact a well defined term (other than phase change). Strictly spoken, only FIR **filter** with symmetrical **impulse** **response** expose linear phase (respectively constant group delay). CIC decimators belong to this class of FIR **filter**s.

PLD, SPLD, GAL, CPLD, FPGA Design :: 10-26-2012 13:34 :: FvM :: Replies: **5** :: Views: **782**

Hi,
Indeed, each bin of the DFT is the result (at one specific time) of the output of the input sequence applied to a FIR **filter**.
The "**impulse** **response**" of the **filter** associated to the k-th bin is a complex exponential that has k cycles in the duration of the data. Its frequency **response** is maximum at the (...)

Digital Signal Processing :: 08-18-2012 12:24 :: zorro :: Replies: **4** :: Views: **10198**

i am using matlab to analyse audio signals .i am using r2008a student version and don't have many options for plotting the frequency i ended up with 'freqz' function which plots magnitude (dB) vs normalized frequency.but during **filter**ing i need to know the actual frequency range i am **filter**ing.say i need to **filter** 700 hz to 1600 hz .how

Digital Signal Processing :: 08-06-2012 04:06 :: ritusubash :: Replies: **1** :: Views: **1254**

One more thing: I find that the **impulse** **response** of a bandpass **filter** seems to be differing depending on the damping coefficient. If we have damping coefficient less than 1 then we will be getting sine waves in the **response** right?
In addition, also the **impulse** **response** of high- and (...)

Analog Circuit Design :: 06-17-2012 18:33 :: FvM :: Replies: **11** :: Views: **1775**

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