What am I missing normalizing a FIR filter?

Started by in comp.dsp16 years ago 10 replies

I analytically derive a simple raised cosine filter with a center frequency and width by taking the inverse transform of the idealized profile...

I analytically derive a simple raised cosine filter with a center frequency and width by taking the inverse transform of the idealized profile in spectral space. The transformed analytical function is normalized so the maximum amplitude is one. It is then digitized and truncated at one of the zeros. The mean of the truncated filter is then subtracted from each element. I apply this filter...

Seeking good technique for preloading an FIR filter

Started by in comp.dsp16 years ago 5 replies

I've got a oversampled signal which is then run through a high order filter. I've set up my DSP program to take advantage of sub-band filtering....

I've got a oversampled signal which is then run through a high order filter. I've set up my DSP program to take advantage of sub-band filtering. One problem though is that I am losing the intitial N samples (N is the filter length) because I am starting my filtering in a steady-state condition - there are a maximum number of samples being multiplied by filter coefficients. The missed s...

interpolation at data sequence endpoints

Started by in comp.dsp16 years ago 3 replies

In a previous thread, it was mentioned that polynomial spline interpolation and windowed sinc reconstruction give similar results, or may even...

In a previous thread, it was mentioned that polynomial spline interpolation and windowed sinc reconstruction give similar results, or may even converge under some conditions. In another recent thread, it was stated the a windowed sinc reconstruction filter used for upsampling would give "invalid" results near the endpoints of a data sequence until all the taps of the FIR filter were filled...

FIR roots and frequency response

Started by in comp.dsp16 years ago 73 replies

I understand that the magnitude of the frequency response of an FIR filter is the intersection of a tube going down through the unit circle...

I understand that the magnitude of the frequency response of an FIR filter is the intersection of a tube going down through the unit circle with the curvaceous surface created by the placement of the zeros of its roots. Is there also a simple visualization of the phase of the frequency response that's related to the zero placement? Bob -- "Things should be described as simpl...

Invariable FIR

Started by in comp.dsp16 years ago 2 replies

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at...

Hello, I am looking for a way to keep a FIR the same using to different sampling rate. I explain : I havean an incoming signal sampling at 1048 Hz in which i want to suppress everything above 50 Hz so i make a low-pass FIR filter (i know a0, a1,.... filter coefficient). But i want to keep the same filter behaviour with the same signal but sampling this time at 10008 Hz for example. ...

Frequency response of an FIR filter

Started by in comp.dsp16 years ago 4 replies

Hi all, If h is the impulse response of an FIR filter then Sum[h(n) Exp(j w n)] describes its frequency response. What is the range of n for...

Hi all, If h is the impulse response of an FIR filter then Sum[h(n) Exp(j w n)] describes its frequency response. What is the range of n for an odd length filter? Is it n = -N : N, or n = 0 : 2N? How about even-length filter; will it be n = 0 : 2N-1 or n = -N/2 : N/2? Regards, Ishtiaq.

Mirror Image of Magnitude Response

Started by in comp.dsp16 years ago 22 replies

I have a real impulse response h of an FIR filter, and its frequency response H is pure imaginary (if I rotate h to left by half of its...

I have a real impulse response h of an FIR filter, and its frequency response H is pure imaginary (if I rotate h to left by half of its length). I want the impulse response of a filter whose magnitude response is 1-Abs[H]. Is it possible to do this with some trick on h? Regards, Ishtiaq.

FIR filter design (false advertizing?)

Started by in comp.dsp16 years ago 18 replies

Trying to design an FIR Low pass filter. As a simple example, consider: > > edge = [0 .3 .7 1]; > > weight = [100 1]; > > ...

Trying to design an FIR Low pass filter. As a simple example, consider: > > edge = [0 .3 .7 1]; > > weight = [100 1]; > > 1000*remez(7,edge,[1 1 0 0],weight) ans= 32.2690 -109.2208 81.9170 495.0689 495.0689 81.9170 -109.2208 32.2690 The resulting filter has a pass band ripple of +/- .013dB and a stop band of -16.84dB. (The pass band is from DC to .3 and the stop band is from .7 to 1 where

Converting a Digital Filter to an Analog Filter

Started by in comp.dsp16 years ago 7 replies

Has anyone heard of a method to convert a digital FIR filter back into an analog filter? Can you perform a bilinear or impulse...

Has anyone heard of a method to convert a digital FIR filter back into an analog filter? Can you perform a bilinear or impulse invariant transform backwards? Thanks Brett Emsley

Is there a point to theoretical understanding?

Started by in comp.dsp16 years ago 42 replies

Greetings: I have begun to learn the subject of DSP, which started out with implementing a working 100-tap FIR filter in an AVR a few weeks...

Greetings: I have begun to learn the subject of DSP, which started out with implementing a working 100-tap FIR filter in an AVR a few weeks ago. This is a subject that I find fascinating, particularly the applied mathematics aspect. But also the hardware aspect, as there is something very satisfying about constructing a real machine that actualizes a mathematical concept. I am pla...

REMEZ FIR filter generation (FIR Design Algorithms)

Started by in comp.dsp16 years ago 1 reply

Hello, i am looking for an algorithme in C in order to generate Parks-McClellan FIR filter. I found one...

Hello, i am looking for an algorithme in C in order to generate Parks-McClellan FIR filter. I found one at "http://www.menne-biomed.de/". I want to know if someone has already use this type of algorithme ? Is there another available ? Is generated FIR coeficients are good ? thanks for your help

Minimum phase filter design using cepstrum methods?

Started by in comp.dsp16 years ago 13 replies

I've run across a few web pages describing how to convert an arbitrary FIR filter to a minimum phase variant by the use of cepstral...

I've run across a few web pages describing how to convert an arbitrary FIR filter to a minimum phase variant by the use of cepstral methods: e.g. > wn = [ones(1,m); 2*ones((n+odd)/2-1,m) ; ones(1-rem(n,2),m); > zeros((n+od d)/2-1,m)]; > y = real(ifft(exp(fft(wn.*real(ifft(log(abs(fft(x))))))))); etc. I've tried this algorithm on a few windowed-sinc filters and it seems to work...

Cancellation

Started by in comp.dsp16 years ago 2 replies

Hi all, I have read up on using echo cancellation and training FIR filter coefficients with various algorithms. All these systems rely on...

Hi all, I have read up on using echo cancellation and training FIR filter coefficients with various algorithms. All these systems rely on two uncorrelated signals to find echo paths. Does anyone know if methods exist to allow echo paths to be found using a single frequency turned on an off or a frequency sweep(chirp) to allow delays to be determined? Could a FIR filter be trained using suc...

FIR coefficient : is there a software which allows to convert real number coefficients to hexadecimal Q15 ones ?

Started by in comp.dsp16 years ago 5 replies

Hi, Here is another question : I'm designing a FIR filter program (on a TMS320VC5402) and as a consequence i calculated the coefficients...

Hi, Here is another question : I'm designing a FIR filter program (on a TMS320VC5402) and as a consequence i calculated the coefficients with an online software. The issue is that this software gives the coefficient in real number representation. In the program, when defining the buffer which contains the coefficients, i guess they must be declared as Q15 numbers or another hexadecimal ...

sferic detection and localisation with DSP

Started by in comp.dsp16 years ago 15 replies

Hi, My project is to realise an handheld lightning locator designed around a Digital Signal Processor. From what i learnt about lightning, i...

Hi, My project is to realise an handheld lightning locator designed around a Digital Signal Processor. From what i learnt about lightning, i think i can detect storm by detecting sferic signals. For the detection, i think that just an antenna+ADC+ FIR filter will be enough but for the localisation i don't know how to do. Should i make a FFT? I thought that as the high frequencies of the s...

pole-zero plot with zeros |a|

Started by in comp.dsp16 years ago 11 replies

A simplified two-path fading radio channel can be modelled as a first-order FIR filter, with delay T, and the strength of the direct signal being...

A simplified two-path fading radio channel can be modelled as a first-order FIR filter, with delay T, and the strength of the direct signal being b0 and the strength of the indirect signal is b1. The channel can be equalised by the use of an IIR filter, which has the opposite frequency response. The frequency response is given by: G/(1+az^-1) where G = 1/b0 and a = b1/b0 However, d...

FIR gain

Started by in comp.dsp15 years ago 11 replies

I have an FIR filter with the system equation y[n] = 0.25x[n]+0.5x[n-1]+0.25x[n-2] which gives an impulse response of h = [0.25 0.5...

I have an FIR filter with the system equation y[n] = 0.25x[n]+0.5x[n-1]+0.25x[n-2] which gives an impulse response of h = [0.25 0.5 0.25] without resorting to Z-transform analysis, how can I work out the gain of the filter at DC and at the half nyquist frequency? I think the gain of the filter at DC is the sum of the impulse response, in this case 1. but what about the half-nyquist? ...

The way to do FIR filter.

Started by in comp.dsp15 years ago 5 replies

I found there are two method to do FIR. A. find the coefficient from fourier series. B. Window method. My thinking about A. 1. Use...

I found there are two method to do FIR. A. find the coefficient from fourier series. B. Window method. My thinking about A. 1. Use fourier series to contribute a desire filter curv(ac response) 2. find fourier series 3. convolution signal with series 4. Get the FIR. I am correct? Thank you very much. Best regards, Boki.

Designing a FIR pre-emphasis filter

Started by in comp.dsp15 years ago 8 replies

I'm attempting to create a FIR pre-emphasis filter. Here's the criteria: sampling frequency = ~100KHz pre-emphasis curve = 75uS from 0 to...

I'm attempting to create a FIR pre-emphasis filter. Here's the criteria: sampling frequency = ~100KHz pre-emphasis curve = 75uS from 0 to 15KHz. As little gain as possible from 15KHz above. I think the application is obvious ;) I have a FIR filter design on my desk which does an excellent job of pre-emphasis as well as attenuation after 15kHz - however, the filter was designed for a ...

Group Delay of Polyphase FIR filters

Started by in comp.dsp15 years ago 3 replies

Could anyone explain or point me to any literature that explains how to compute the group delays of polyphase FIR filters used for decimation...

Could anyone explain or point me to any literature that explains how to compute the group delays of polyphase FIR filters used for decimation and interpolation? For example if an FIR filter with (N*L) coefficients is implemented as L polyphase filters each with N coefficients, what is the group delay for the filter? Measured values indicate the group delay to be less than (N*L)/2 which is ...