Interchanging FIR and Decimation operation

Started by tara...@gmail.com in comp.dsp14 years ago 13 replies

Hi all, I have a yet another decimation question rather a decimation and a FIR filter question. I have 64K input samples that are filtered...

Hi all, I have a yet another decimation question rather a decimation and a FIR filter question. I have 64K input samples that are filtered using a 3 tap - 2nd order FIR filter, say [a1 a2 a3]. After filtering they are decimated by 8. I want to reverse this operation such that I decimate first and then filter it for obvious reasons (why should I compute the taps that anyways I am going ...


FIR notch filters using the Windowing method

Started by Greer in comp.dsp14 years ago 41 replies

Hi, I am trying to obtain a FIR notch filter which is very narrow. For example, a 50 Hz notch filter to remove mains noise from a 1250Hz...

Hi, I am trying to obtain a FIR notch filter which is very narrow. For example, a 50 Hz notch filter to remove mains noise from a 1250Hz signal say. Doing this with an IIR filter was easy, I could specify my "notch region" as small as [49 51]Hz. I am now trying to do the same with a FIR filter. I am using Intel Peformance Primative functions to generate the taps. These use the windowing ...


Hilbert transform question

Started by Rick Lyons in comp.dsp14 years ago 16 replies

Hi Guys, I have a question about Hilbert transformer applications. First, we can build Hilbert transformers using a tapped-delay line...

Hi Guys, I have a question about Hilbert transformer applications. First, we can build Hilbert transformers using a tapped-delay line structure (like a tapped-delay line FIR filter.) An ideal Hilbert transformers (HT) would have a freq magnitude response that's flat over the HT's entire operating frequency range. However, practical HTs have a magnitude null at zero Hz and a...


FIR Coefficients

Started by Himanshu Chauhan in comp.dsp14 years ago 20 replies

Hello, I have been working on a low-pass FIR filter using Hamming and Blackman window. What I have is a function named "get_filter_kernel"...

Hello, I have been working on a low-pass FIR filter using Hamming and Blackman window. What I have is a function named "get_filter_kernel" which calculates the filter kernel. Now, If I wish to implement a FIR interpolator, can I use this filter kernel for my purpose. Are the filter kernel and Filter coefficients one and the same thing? Thanks and regards --Himanshu ============...


Interpolated frequency sampling linear-phase FIR design

Started by Andor in comp.dsp14 years ago 8 replies

Friends, I face the following problem. Because several vectors of differing lenghts occur in the description, I use Harris' convention of...

Friends, I face the following problem. Because several vectors of differing lenghts occur in the description, I use Harris' convention of writing [h(n):M] to denote the vector h of length M. Assume I specify the desired magnitude response of a linear-phase FIR filter at N/2+1 evenly spaced points [H(n):N/2+1], starting with 0, ending at Fs/2, where Fs is the sampling rate. Then use the i...


FIR Filter: Silence in between

Started by Himanshu Chauhan in comp.dsp14 years ago 13 replies

Hello! I have written a FIR Filter (High Pass, low pass). Everything is working as expected except one. Its 100 tap filter so after...

Hello! I have written a FIR Filter (High Pass, low pass). Everything is working as expected except one. Its 100 tap filter so after convolution there is a silence of 100 samples in output data. Input data is a wav file (44100 Hz, 16 Bit, Mono). High Pass perfectly cuts down the low frequencies but silience (in the begining of every buffer) remains which produces clicks in the final file...


How to Convert 8-bit data to 10-bit data?

Started by Anonymous in comp.dsp14 years ago 4 replies

Hi, I have the following question. How to convert an 8-bit data to 10-bit data ? The following C program uses 8-bit data. (Basically FIR...

Hi, I have the following question. How to convert an 8-bit data to 10-bit data ? The following C program uses 8-bit data. (Basically FIR filter) /*********************************************************/ #define MAX 6 typedef unsigned char uint_8 uint_8 fir() { uint_8 buffer[MAX], out; out =( (buffer[0] + buffer[5]) + ((buffer[1] + buffer[4]) * 2) + ...


CIC Filter design for multiple FM carrier demod

Started by Paul Solomon in comp.dsp14 years ago 22 replies

Hi All, I am working on a project in which we are attempting to demod multiple (analog) FM radio stations in a FPGA. I have been trying to...

Hi All, I am working on a project in which we are attempting to demod multiple (analog) FM radio stations in a FPGA. I have been trying to work out how to design the CIC / FIR filter pair in the DDC section of this design. I have a input sampling rate of 80MSPS, which undersamples a clean spectrum of 88 - 108MHz i.e. the FM band. this should give me the FM band at 8 - 28MHz with an a...


How can I calculate group delay of a FIR filter on PC?

Started by Anonymous in comp.dsp14 years ago 9 replies

I am trying to calculate group delay of a FIR filter on PC. I use Matlab but I am confused with its algorithm. Basically the Matlab algorithm...

I am trying to calculate group delay of a FIR filter on PC. I use Matlab but I am confused with its algorithm. Basically the Matlab algorithm says this, group_delay(f) = Fourier transform of {n * h[n]} where h[n] is impulse response of the filter. But the Fourier transform of h[n]*n is the differentiation of H(w) with respect to w and not differentiation of phase of H(w) with respect to ...


FIR filter phase shifter

Started by jim_dsp_q in comp.dsp14 years ago 3 replies

I'm trying to generate sets of all-pass FIR filter coefficients to apply varying degrees of phase shift (say from 0 to pi/2) to filter...

I'm trying to generate sets of all-pass FIR filter coefficients to apply varying degrees of phase shift (say from 0 to pi/2) to filter input signals. Can anyone suggest a good way to go about calculating these? Will it be similar to the derivation of a Hilbert Transform? Thanks very much, Jim Singleton This message was sent using the Comp.DSP web interface on www.DSPRelated.com


Make a FIR filter remove the DC component

Started by Rune Allnor in comp.dsp14 years ago 7 replies

Hi all. I am making these bandpass filters to enhance some low-frequency data. The pass band is specified as [0.009, 0.015]*Fs. The...

Hi all. I am making these bandpass filters to enhance some low-frequency data. The pass band is specified as [0.009, 0.015]*Fs. The filters are of order > 400 and work reasonably well, except for there being some non-zero gain at DC. Can I play with the filter order to remove the DC component? I know I can remove a non-zero component at Fs/2 by choosing an odd filter order (which, app


Understanding IIR filters -- Audacity's Bass Boost

Started by Himanshu Chauhan in comp.dsp14 years ago 3 replies

Hello! I am learning about iir filters. For this I was going through the bass boost effect of audacity. Can anybody please tell me how exactly...

Hello! I am learning about iir filters. For this I was going through the bass boost effect of audacity. Can anybody please tell me how exactly the filter coefficients are being calculated? I wonder how do we come to a point of understanding that "this" algorithm for calculating the coefficients is the one for bass boost or any other effect. I tried applying the same effect with FIR filter...


optimal BPSK receiver with Raised Cosine

Started by khubaib in comp.dsp14 years ago 1 reply

Hi, I have a practical problem, which I want to discuss with you. I have a baseband signal, which is BPSK modulated with a pulse shape as :...

Hi, I have a practical problem, which I want to discuss with you. I have a baseband signal, which is BPSK modulated with a pulse shape as : alternative polariy pulse for 1s and zeros for 0s. My transmitter employs a wave shaping filter (which is a RC filter). The receiver starts from a first an FIR low pass filter, then a matched filter (which is an FIR filter derived from MATLAB's multi-ban...


FIR filter in Matlab

Started by TheAmbison in comp.dsp14 years ago 1 reply

Greetings, I am a student from Vienna, Austria and working on my bachelor of arts research project. I am testing the effect of shelf filtering...

Greetings, I am a student from Vienna, Austria and working on my bachelor of arts research project. I am testing the effect of shelf filtering in an specialized surround sound system on the sonic image. ok, you don't have to know what that means :) But i have a problem. I need the FIR filters (because i got a lot of hints that they would work a lot better) in MATLAB format but i never l...


newbie: fir filter design

Started by Anonymous in comp.dsp14 years ago 17 replies

Hi again, I need to design a FIR filter with an even number of coefficients. I was wondering if the following formula is correct for...

Hi again, I need to design a FIR filter with an even number of coefficients. I was wondering if the following formula is correct for calculating the coefficients for a lowpass filter with hamming window - for(n=0;n


90 degrees phase shift

Started by Giuseppe Sbarra in comp.dsp14 years ago 24 replies

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the...

Hi, I'm an hold analog engineer , actually developing a DSP based application and I need to change of 90 degrees the phase of a signal on the range of 50 - 250 Hz aving the system a 200uSec sampling rate. I have considered the Hilbert FIR filter but for the moment I cannot get it to work not even reducing the sampling rate. In particular I nedd to phase shift by 90 degrees a signal (voltage)...


circular FIR filter

Started by Anonymous in comp.dsp14 years ago 5 replies

If I have a 1024 tap filter and I wish to use a circular FIR filter to process 256 samples for each call of the associated function, then...

If I have a 1024 tap filter and I wish to use a circular FIR filter to process 256 samples for each call of the associated function, then what is the minimum size that I need my circular buffer to be?


FIR & IIR stabilitity

Started by Sarah in comp.dsp14 years ago 3 replies

Hello all, I have an FIR filter (h) which I convolve (filter) with a sequence of data (x) that is: y=x*h; in matlab, I use:...

Hello all, I have an FIR filter (h) which I convolve (filter) with a sequence of data (x) that is: y=x*h; in matlab, I use: y=filter(h,1,x); Now, I want to recover the original data from the convolved (filtered) sequence (y), and in matlab I use: x1=filter(1,h,y); but in the reverse process, the original FIR filter becomes, IIR and unstable and I can't recover the original data. now ...


3 FIR filters

Started by athiq006 in comp.dsp14 years ago 8 replies

hello i was given a project that uses three DC centered FIR filters. the ouput power of three FIR filter is used to compute the fine...

hello i was given a project that uses three DC centered FIR filters. the ouput power of three FIR filter is used to compute the fine filter split on [-1/2,1/2]. I wanted to know what this fine filter split means. and how to calculate the interepolation coefficients. -rahman This message was sent using the Comp.DSP web interface on www.DSPRelated.com


FIR filtering in the Fourier domain

Started by Philip de Groot in comp.dsp14 years ago 57 replies

Hello, I am aware that many questions regarding FIR-filtering are posted here and many links to websites to help people out are also...

Hello, I am aware that many questions regarding FIR-filtering are posted here and many links to websites to help people out are also included. But these links are becoming complicated and I want to know something simple. Therefore, I am asking. I have a Fourier transform and want to remove frequencies below 600 Hz and above 6500 Hz and apply a IDFT. I can use a FIR filter (in the fre...