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37 Threads found on edaboard.com: **Inverse Filter**

Hello,
suppose there is a bessel **filter** (2nd order low pass) which has the following transfer function:
TF_{Bessel}(s) = \frac{a_1}{a_1+b_1s+s^2} .
I apply an input voltage of the form:
V_{in}(s) = \frac{V}{s}
I would like to find an anlytic description of the bessel output.
Calculating the **inverse** laplace transform

Digital Signal Processing :: 03-04-2017 11:09 :: twig27 :: Replies: **0** :: Views: **591**

Hello all, I am having a bit of trouble trying to implement this system in the figure to first have a binary signal go through a channel (which i've done) I then don't know what to do to get the new signal back to binary and how to do the training and decision directed parts. I am struggling to write the code in matlab to implement this.
I have t

Digital Signal Processing :: 01-03-2017 18:36 :: Poley :: Replies: **0** :: Views: **625**

The cepstrum is defined as the **inverse** Fourier transform of the log-magnitude Fourier spectrum. The Mel scale is roughly linear with Hertz scale to 1kHz then with increasing spacing approx. 50% per octave above this, according to human hearing perception.
The MFCC scaled **filter** banks then become wider at higher frequency vs equal spaced for li

Digital Signal Processing :: 08-15-2015 14:09 :: SunnySkyguy :: Replies: **2** :: Views: **1023**

i wanted to know how FIR LPF **filter** are derived
To get a basic idea, you can try a classical design method, called "direct synthesis":
- perform an **inverse** fourier transform of the intended frequency response. Assume a pure real-valued function for simplicity.
- apply a window (e.g. Hanning, Hamming, Blackman) to achieve a smoot

Digital Signal Processing :: 03-31-2015 18:43 :: FvM :: Replies: **9** :: Views: **998**

Synopsis use something like this
The pattern has the centroid **inverse** weight (1,2) with guarding (3) and sidebands 4~9
The weighted average improves accuracy from 10 to 12 bits and reduced size improves speed into microwave DAC speeds.
It is like decimating a 1 D **filter** into 2D

Analog Circuit Design :: 03-18-2015 21:11 :: SunnySkyguy :: Replies: **4** :: Views: **1689**

You'll remove the "echo" in the transfer function by a **filter** with the **inverse** transfer function.
The echo cancellation **filter** will be however an IIR **filter**.

Digital Signal Processing :: 10-17-2014 09:11 :: FvM :: Replies: **7** :: Views: **1287**

max distance with minimum power can be traded off with speed of data and modulation.
IR wavelength does not matter much as long as receiver bandwidth includes transmitter and you have a daylight blocking **filter**.
Path loss is **inverse** squared, so smallest beam angle in emitter is desirable
Power per bit can be computed as VI*t from transmitt

Digital Signal Processing :: 09-08-2014 23:39 :: SunnySkyguy :: Replies: **1** :: Views: **469**

The fast explanation is that the FFT is a cyclic convolution. For example, people sometimes try to **filter** data by taking an N-point FFT adjusting coefficients, and then doing the **inverse** FFT. This would be nice if it always worked. But even a simple problem, like modeling a sample delay, is impossible. The FFT takes in N samples, and outputs N

Digital communication :: 03-12-2014 07:40 :: permute :: Replies: **4** :: Views: **875**

hello..
i need basic idea about predisotortion...i m going to work with low pass **filter** at the input side and output side bandpass **filter** ..maily in my stsytem contains random intereger generator,qpsk modulato ,raise cosine **filter** then i waill apply awgn channel after that block i want to make response linear so i will add bandpass (...)

RF, Microwave, Antennas and Optics :: 02-08-2014 17:26 :: Bhavik Patel :: Replies: **0** :: Views: **1209**

If you know how to do FFT in excel, then you know how to do the **inverse**-just select the "**inverse**" button in the dialog box.

Software Links :: 01-13-2014 12:56 :: barry :: Replies: **8** :: Views: **85**

Hello Friends.
Please help. I try to learn another's code, that had been written in assembler for ADSP 21062 digital signal processor. In this program I have fliter coeffs.
How I can do **inverse** conversion????? Is it realy to get **filter** characteristics (Amplitude-freq characteristic, Phase-freq characteristic) by the **filter** coeffs????????

Digital Signal Processing :: 08-20-2013 12:15 :: gopyan :: Replies: **0** :: Views: **505**

Spectrum lab is excellent!
I can see there is a plugin to demodulate I/Q. I have not used it extensively but I hope you could connect directly the outputs of an image reject mixer to the stereo input of a PC soundcard to recover and **filter** the audio.
I would like to ask you if there is any plugin for the **inverse** process, i.e to create an audio

RF, Microwave, Antennas and Optics :: 03-17-2012 09:50 :: neazoi :: Replies: **0** :: Views: **1857**

Hi, i need to protect a circuit input from **inverse**r polarity feed without any meaningful voltage drop. I was thinking on the basic parallel TVS diode with a fuse and **filter** capacitor so i will get an extra surge protection and **filter**ing. At a normal operation the circuit will powered by a 12V supply with a peak current of 14A, but a (...)

Analog Circuit Design :: 02-06-2012 00:00 :: Sink0 :: Replies: **0** :: Views: **819**

I wish to know about pseudo **filter** and pseudo **inverse** **filter**..

Elementary Electronic Questions :: 06-22-2011 09:49 :: devika v kurup :: Replies: **0** :: Views: **1056**

I wonder how you want to apply the pwm modulated voltage to the ADC input, it would require a rather slow **filter** to achieve this without considerable errors. I think, that a programmable resistive voltage divider is a more promising option.
To reduce the offset of a BJT operated as analog switch, you can connect it in **inverse** mode, but the curre

Microcontrollers :: 10-08-2010 15:52 :: FvM :: Replies: **2** :: Views: **1232**

A classical design method is based on the **inverse** fourier transform of the frequency repsonse, multiplied with a window function. I think, it's called "direct" synthesis or something similar.

Digital Signal Processing :: 09-16-2010 08:58 :: FvM :: Replies: **6** :: Views: **877**

Refer the attached text for example of IIR **inverse** **filter** (Ex 5.5: IIR**inverse**)

Digital Signal Processing :: 07-21-2010 16:10 :: bassa :: Replies: **4** :: Views: **3238**

Hi,
**inverse** chebyshev passive **filter** are also called type ii chebyshev **filter**. plz check following it may help you,
Digital Chebyshev **filter** Design help
Chebyshev Type II **filter** order - MATLAB

Analog Circuit Design :: 07-19-2010 07:22 :: hanif :: Replies: **1** :: Views: **1925**

So I'm doing some digital signal processing. I have a a current and a voltage signal that I am multiplying together and then **filter**ing through a lowpass **filter** to get a signal that equals A*B*cos(phi)/2. At this point I want to find phi, so I need the **inverse** cosine function.
Unfortunatly I'm having troubles trying to write a cordic (...)

Digital Signal Processing :: 06-18-2010 21:56 :: rawbus :: Replies: **3** :: Views: **2251**

Lock time is **inverse** proportional to PLL **filter** bandwidth.
You should 2 things simultaneously
-Locking time
-Bandwidth
Also, you have to find a optimum way between trade-off

RF, Microwave, Antennas and Optics :: 03-13-2010 21:08 :: BigBoss :: Replies: **5** :: Views: **1195**

Good Morning Forum,
i am a beginner in signal processing, so keep it easy on me. I have an transfere function of a system and i want to build an preconditioning **filter** that ereases the linear system. So I have heard that I can use the invererted transfere function.
code in matlab?=
invH = (H)^(-1); % making the **inverse**
invh = ifft(invH

Digital Signal Processing :: 11-23-2009 09:21 :: Distortion-andi :: Replies: **0** :: Views: **699**

Hi
i am trying to **filter** a data set in matrix format using gaussian **filter**.i take the fourier transform of the data and then multiply with gaussian **filter** and take the **inverse** fourier transform of that. Now here is the function i have written(matlab) for gaussian **filter** design but seems to be not (...)

Digital Signal Processing :: 07-18-2009 14:18 :: tanzil_dhk :: Replies: **0** :: Views: **1290**

nikhil_bhadani,
I would use a Digital Signal Processor (DSP) and Implement the Kalman algorithm in software. Implementing it in hardware would be a daunting task. Most DSPs have single instructions for matrix manipulation such as multiply, **inverse**, transpose, etc.
Regards,
Kral

Digital Signal Processing :: 02-03-2009 14:29 :: Kral :: Replies: **2** :: Views: **1158**

i need to design a **filter** (any iir fir) with linear transition area(pass band to stop band).....
and also a high pass **filter** with the same......
any ideas...books what not....
Another thing......If a **filter** removes or attenuate (low attenuation) unwanted components.....how can i put it back...or amplify IT there an (...)

Digital Signal Processing :: 02-27-2008 10:38 :: roykyn :: Replies: **7** :: Views: **1769**

Hi,
I have an image whose fourier transform is plotted. Now, my project is more related to optics of the eye. So, I have to induce a "defocus" by implementing a bessel **filter** and then the usual process (transform of **filter** * transform of image and then taking **inverse** transform to get the defocussed image). By defocussed, i mean, the image (...)

Elementary Electronic Questions :: 10-01-2007 11:53 :: cedance :: Replies: **0** :: Views: **933**

The RLS (recursive least squares) algorithm is another algorithm for determining the coefficients of an adaptive **filter**. In contrast to the LMS algorithm, the RLS algorithm uses information from all past input samples (and not only from the current tap-input samples) to estimate the (**inverse** of the) autocorrelation matrix of the input vector
bot

Digital Signal Processing :: 08-06-2007 03:21 :: nagacnu :: Replies: **1** :: Views: **1371**

I am currently designing a LC ladder **filter**.
I have already derived the **filter** transfer function from matlab.
It's 5th order **inverse**-chebyshev **filter**.
However, I have no idea how to implement it with LC ladder **filter**.
what's the cirucit topology?
How much the L and C value?
(By the way, I want to (...)

Analog Circuit Design :: 10-26-2006 13:35 :: Lantis :: Replies: **1** :: Views: **1135**

hey...How to use S-function in matlab??
--------------------------------------------------------------------------------
I need to know how to use S-function in matlab to design a spacific **filter** with (**inverse** matrix) as transfer function???

Software Problems, Hints and Reviews :: 06-23-2006 16:44 :: is_shaalan :: Replies: **0** :: Views: **1083**

I need to know how to use S-function in matlab to design a spacific **filter** with (**inverse** matrix) as transfer function???
Added after 2 hours 4 minutes:
I need to know how to use S-function in matlab to design a spacific **filter** with (**inverse** matrix) as transfer function???
Please I ne

Software Problems, Hints and Reviews :: 06-19-2006 09:46 :: shaalan :: Replies: **0** :: Views: **3604**

solve your circuit in the s-domain
get the the transfer function in s-domain
then take **inverse** laplace
you 'll get the time domain

Analog Circuit Design :: 05-15-2006 06:45 :: relqueseny :: Replies: **1** :: Views: **586**

ihave written a MATLAB code for a hilbert transformer :
first a half band **filter** is generated by sampling from raised cosine function in frequency domain and taking **inverse** fourie transformer,
the hilbert transform is computed by subtracting .5 from transfer function and modulating the signal by e^jnpi/2 in the time domain and added with half of

Digital Signal Processing :: 03-22-2006 10:25 :: omidi_sbu :: Replies: **0** :: Views: **1748**

hey i want to know what do you mean by spectrum of the residual signal is flat in speech context .
to eloborate if i have a speech signal and if i perform LPC analysis on it to get the **inverse** filer, and i fiter the speech signal through this **filter** i will get a a residual which is periodic(quasi) impulse train for voiced case and a noisy wave

Analog Circuit Design :: 10-17-2005 06:15 :: bhupala :: Replies: **0** :: Views: **946**

Can any one pleease explain me what are **inverse** models and where they are used

Digital Signal Processing :: 09-04-2005 07:08 :: bilalkadri :: Replies: **1** :: Views: **943**

Hi there.
I am new to DSP and PSoC (Hell, then why I am thinking of doing something!!)
Well, my idea is simple. I have a good turntable with moving magnet pickup. It was connected to Antique Pioneer amp which gave up. Now I plan to hook it to my Sony RV990D DVD system.
First, I though of building a traditional Moving magnet preamp with RIA

Digital Signal Processing :: 01-20-2005 12:21 :: asit :: Replies: **0** :: Views: **1568**

Weighting **filter**s are used in audio S/N measurements. Basic shape of **filter** is **inverse** to human ear sensitivity characteristic. One of these shapes is A weighting **filter** (exists allso B, CCIF etc). Idea is to give more weight to frequency components at which ear is more sensitive. On web you can find a lot data about audio (...)

Analog Circuit Design :: 12-09-2004 08:06 :: Borber :: Replies: **9** :: Views: **5117**

1. Attached pdf will explain all about dB
2. Take in account that for acoustic measurements you must to use
weight **filter** (A- curve, for example) that have frequency response
**inverse** to human ear
2. Log calculation may be too heavy for small processor,
so better to use hardware conversion - log amplifier (from Analog or TI)

Elementary Electronic Questions :: 11-30-2004 17:34 :: jourval :: Replies: **2** :: Views: **2605**

It's hard to say what's the structure inside AD6633.
But after seen the datasheet,you can find there are a error **filter** which use to simulate the channel **filter** response,and small nco which simulate nco I think all this used to generate an **inverse** signal to clip the peak value after NCO combination.
I believe this all are very similar

Digital Signal Processing :: 06-11-2004 11:41 :: blueteeth :: Replies: **1** :: Views: **1168**

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