**just like half band filters are ther 1/3 band filters too?**Hello,
Half band filters are good for interpolation and decimation by
2 as they have alternating 0's as their coefficients.
Is there something like 1/3rd band filters for doing interpolation
and decimation by 3 with in-between 2 coeffcients being 0?
Regards
Bharat
On 2/2/2011 6:16 AM, bharat pathak wrote:
> Hello,
>
> Half band filters are good for interpolation and decimation by
> 2 as they have alternating 0's as their coefficients.
>
> Is there something like 1/3rd band filters for doing interpolation
> ...

**i need to design low pass , high pass, bandpass filtr with cut off frequency 6,18,20 khz. pls guide me**when i type these functions in matlab , i get default fstop, astop, etc. how could i correlate these things with that of cut off frequency and gain of filter. pls help me.....
"nethaji anandhavalli" <mydreamprojects@yahoo.com> wrote in message <hq7fu8$rbb$1@fred.mathworks.com>...
> when i type these functions in matlab , i get default fstop, astop, etc. how could i correlate these things with that of cut off frequency and gain of filter. pls help me.....
Hi Nethaji, I assume you are talking about using a filter specification object. You do not state in your post ...

**Narrow pass band filter design**Dear all,
This is in ref. to below link:
http://www.mathworks.com/support/solutions/en/data/1-16LFT/?solution=1-16LFT
In above link, they mention 0.012 as very small. What is the threshold value required for ratio (2*fc/fs) to design a stable IIR Butterworth low pass filter? Here fc is cutoff frequency & fs is sampling frequency.
Regards,
abhishek
"Abhishek Ballaney" <denim83@yahoo.com> wrote in message <igk010$pv4$1@fred.mathworks.com>...
> Dear all,
>
> This is in ref. to below link:
> http://www.mathworks.com/support/solutions/en/dat...

**How can i design a band-pass filter with scipy?**Hi all,
i would like to design a high pass filter with scipy.signal module.
This is the code i'm using to:
import scipy.signal as signal
import scipy
#first of all i design the lowpass fir filter. This is a 10 taps filter
with cutoff frequency =1 (as help tell me to do)
lpwindow=signal.firwin(10,1)
#with the following instruction i'm creating a band pass filter from
the low pass one
bpwindow=signal.lp2bp(lpwindow,1,0.5,0.2)
My problem is that the band-pass filter obtained with lp2bp function is
16 taps one!
How is it possible??
thanks,
Vincenzo
LabWINC wrote:
> Hi all,
>...

**IIR band pass filter design query**Hi,
i have to design a digital Band pass filter using butter worth
4th order filter.... the requirement is that the pass band frequency i
1KHz and the sampling frequency is 48KHz. the filter should have a ver
high Q factor...no band is specified as such
my question is
1. Q factor is related to analog filter and how is it translated to
digital filter design
2. what is the best Q factor that could be got and how to measure it
3. any matlab functions to implement it like butter()
any other related material would be good...
thanks
--srikk
On Aug 29, 8:03 am, "srikk123...

**Low pass filter design with complex-signal input**Hi,
I am recently trying to use low-pass filter under Simulink. My input signal is complex-valued which is BPSK with PN-sequence after AWGN channel, but I've tried "FDA tool" and lowpass filter and RF lowpass filter, the error message are all the same, as "Complex signal mismatch. Input port 1 of 'testfile/Lowpass RF Filter/RF Filter/Out' expects a signal of numeric type real. However, it is driven by a signal of numeric type complex" like that.
How can I deal with this problem? I think there should be somewhere parameter setting under the lowpass filter ...

**Signal processing; Fourier Transformation and Band Pass Filtering**Hello All;
I have a homework; in signal processing,
What i need is that i ha ve a data like;
ID miliseconds microvolts
0 -1024.00 -.2811
1 -1022.00 -.3166
2 -1020.00 -.0769
3 -1018.00 -.2151
4 -1016.00 -.3521
5 -1014.00 -.5004
6 -1012.00 -.3115
7 -1010.00 -.2304
8 -1008.00 -.2887
9 -1006.00 -.1213
......
1016 1008.00 -.1873
1017 1010.00 -.1467
1018 ...

**How to apply 1/3 octave bandpass filter of about 10Khz on a signal**I would like to know if octave filtering is possible in Labview.
There is a toolkit for octave analysis. <a
href="http://sine.ni.com/apps/we/nioc.vp?cid=3122&lang=US">NI LabVIEW
Sound and Vibration Toolset</a>
Hello Pavitra,
The Sound and Vibration Toolset has many powerful tools for octave
analysis. Please refer to chapter 8 of the <a
href="http://digital.ni.com/manuals.nsf/webAdvsearch/C31B0809E0EE1934862567B5004F30F6?OpenDocument&vid=niwc&node=132100_US">Sound
and Vibration Toolset Reference Manual</a> for more information on
these...

**How to apply 1/3 octave bandpass filter of about 10Khz on a signal #2**I am unable to retrieve the time signal after doing the 1/3 octave
analysis.
Can you post a simple example program?
...

**your intuition: 16MHz MIPS for tri-band SSB Demod at 44.1 KHz???**
Hi! to expand on the subject briefly: I'm experimenting with mechanical
television a la Baird televisor. I think it would be neat to encode
three channels of color (YIQ) into one audio channel. I would like
to use a fairly cheap microcontroller for demodulation because I have
lots of experience with it (Atmel Atmega32).
Thus far, SSB AM has been the most straightforward approach. One of
its advantages is that I can place the Y channel in the highest
frequencies (e.g. from 8-22 KHz) and if intermediate equipment
is lossy, it will only mean a loss of high frequency data.
However, a...

**on extracting band-pass signal by using Goertzel algorithm instead of DFT filter banks** On the poly-phase filter banks, it's straightforward to use the
Goertzel agorithm instead of DFT can reduce the costs if only one user
is to be extracted. In the Tim Hentschel and Tuttlebee's book and some
related articles, there're some discussions on this topic. The
following link is available, related discussions on p.p. 31.
http://citeseer.ist.psu.edu/hentschel02channelization.html
While I check this algorithm, I found a question. The goertzel
algorithm is a single-pole IIR filter, which means that this filter
has a very narrow frequncy response at the pole bin. This c...

**Designing a second order IIR filter to amplify a desired signal in the presence of low amplitude broad band background noise**I have downloaded a discrete time signal called pcm.mat file in order to amplify it with a second order IIR filter. Sampling rate for the signal is 8kHz. I used SPTool in MatLab to get the Spectra of the signal. I need to determine the frequency of the signal to is to be amplified. One problem, I can't tell and don't know what peak or minima to chose and amplify. Can anyone please help? Thanks.
-Aneel
...

**ANSOFT HFSS V9.2, PEXPRT V5.0, SIMPLORER V6.0, Agilent Vee Pro v7.0, AGILENT T&P Tookit 1.2, IC-CAP V2002, ADVANCED DESIGN SYSTEM V2002C, Agilent 89600 Series Vector Signal Analyzers 3.01a, HFSS**ANSOFT HFSS V9.2, PEXPRT V5.0, SIMPLORER V6.0, Agilent Vee Pro v7.0,
AGILENT T&P Tookit 1.2, IC-CAP V2002, ADVANCED DESIGN SYSTEM V2002C,
Agilent 89600 Series Vector Signal Analyzers 3.01a, HFSS V5.6 -
AGILENT/HP, other
HFSS V9.2 - ANSOFT CD NR 16 109
DESIGNER V1.1 - ANSOFT 14 296
HFSS V9.1 - ANSOFT 14 297
HFSS V9.0 - ANSOFT 12 815
PEXPRT V5.0 - ANSOFT 12 816
ANSOFT Maxwell...

**44.1 khz to 48 khz using polyphase**Hi,
I am trying to do a 44.1 khz to 48 khz resampling in Matlab, but am
coming across a few problems. For this conversion to take place the
poly phase filters come out like
480/441=(4 x 4 x 10)/(7 x 7 x 3). I have been trying to figure this
thing out on paper, like the for the first stage I get
44100 x (4/7), now as I understand it I have to take a 44.1 khz signal
upsample it by 4, apply an FIR filter with cutoff pi/7 to it, and then
downsample it to get 25.2 khz which is the output of the first stage.
But when I plot this thing on paper I get serious aliasing and
overlaps, the same is tru...

**FIR Digital Filter using Simulink (Low Pass, High Pass & Band-Pass FIR digital filters)**Dear All,
I looking for a self made FIR Digital Filter using Simulink.
I would like to know if there is any freely available webinars or links to learn "How to realize a FIR-Digital filter (Low Pass, High Pass & Band-Pass FIR digital filters) in simulink". I know there is a Digital filter block diagram but I want to know If I can design my own Filter in Simulink.
If any one has idea then I will be glad to know.
Thanking you,
With Regards,
Prashant
"Prashant Sandhi" wrote in message <jg8obe$17l$1@newscl01ah.mathworks.com>...
> Dear ...

**Band Pass & Stop Band Filter Coef Transform from Low Pass**Hello,
as I am not an expert I am asking help to transform the IIR Coefficients
of a Low Pass Butterworth 6 poles filter to Band Pass and Stop.
I coded my application in C and after performing the computation of the
normalized coefficient with normalized frequency of 1 Hz, I performed the
low pass to low pass transform and the low pass to high pass transform
obtaining the new coefficients. (To that I used a commercial library but in
it I am not able to find the others transform). Then I performed the
filtering of data with success. After I need to transform the normalized
coefficients to band ...

**C++ and design Pattern (Composite design Pattern ) #3**HI All
i am new to C++ and design Pattern ... while implementing Composite
design Pattern
Facing Issue. Kindly help me Out.
[root@arnav test]# cat Filesystem.cc
# include <iostream>
# include <vector>
# include <string.h>
# include <sstream>
# include <ctime>
using namespace std;
#define PORTAGE_DIR "/home/test/"
# include <stdio.h>
#include <stdlib.h>
# include <dirent.h>
# include <sys/types.h>
# include <sys/stat.h>
///////////////////////////////////////////////////////
class Directory;
class AbsFile {
s...

**How can I design a bandpass filter or a high pass filter?**Hi everyine and thanks for reading my question!
I want little brief examples on how to desing a bandpass filter and a high pass filter in matlab starting from my discussion on here:
http://www.mathworks.com/matlabcentral/newsreader/view_thread/257121
Can you give me little 2 examples for these 2 filters?
"Sprinceana " <mihaispr@yahoo.com> wrote in message <h4mgq7$jmk$1@fred.mathworks.com>...
> Hi everyine and thanks for reading my question!
>
> I want little brief examples on how to desing a bandpass filter and a high pass filter in matlab starting from my d...

**in-band DTMF signalling on Cisco Call Manager 3.3**Anyone know how to set DTMF signaling to be in-band using Call Manager
3.3 and a 7960 IP phone? The phone is using Skinny to talk to the Call Manager.
jim.brunetti@usa.net (Jim Brunetti) wrote in message news:<79fd561f.0407082001.4bb567f8@posting.google.com>...
> Anyone know how to set DTMF signaling to be in-band using Call Manager
> 3.3 and a 7960 IP phone? The phone is using Skinny to talk to the Call Manager.
Jim,
Can you explain more about your requirement? The 7960 signals DTMF
events to the CM via SCCP. The CM then signals this to the PSTN
via the GW via H.245 o...

**Band Pass Filter**I have an image with 2 different surface. one is smooth another is rough... I would like to detect the rough surface...Thought of using a band pass filter to remove the low freq components and high freq noise. then by a thresholding operation and low pass filter...
Anyone has example coding of the operations above?? Thanks in advance.
...

**Filter Design #3**Could anyone give me a short introduction in filter design?
I just want to define a Butterworth filter and apply it on my data
but with
n=5;
[A B] = butter(n,.6,'low');
I get
??? Undefined command/function 'butter'.
Error in ==> Butterworth at 8
[A B] = butter(n,.6,'low');
although I have the signal processing toolbox.
Do I have to include the toolbox in any way in the syntax?
Thanks
Michael Schrefl wrote:
>
>
> Could anyone give me a short introduction in filter design?
> I just want to define a Butterworth filter and apply it on my data
> but w...

**Filter a sinusoidal signal**Hi All..
I'm currently trying to filter a FM signal... the code is below
clear all
close all
echo on
t0=.2; % signal duration
ts=0.001; % sampling interval
fc=250; % carrier frequency
fs=1/ts; % sampling frequency
t=[-t0/2:ts:t0/2]; % time vector
kf=100; % deviation constant
m=cos(2*pi*10*t); % the message signal
int_m(1)=0;
for i=1:length(t)-1 % integral of m
int_m(i+1)=int_m(i)+m...

**filter design #3**Hi,
Suppose I want to design a digital filter for some application. There
are some options to do this:
- weighted least squares design
- minimax design
- filter design based on 'windows' (fir1, fir2)
- equiripple design (remez)
Which method is the most used (in education, in industry)? Which method
results in a filter satisfying the filter requirements the best?
Thanks,
Laki
laki wrote:
> Hi,
>
> Suppose I want to design a digital filter for some application. There
> are some options to do this:
> - weighted least squares design
> - minimax design
> - filte...

**All Pass Filter design**Hi; I'm having a little trouble implementing the all-pass filter. I hav
implemented an all-pass filter, when I run it on the DSK it seems wor
fine; the output sound of the filter is the same as the input (but just
little quieter). However, if I increase R (delay value) to more than 19
samples (=4ms with sample rate of 48khz), i get a really high pitc
ringing effect! Also, when the output of the allpass filter is mixed wit
the original input, there is no phasing effect audible what so ever!
have listed my filter code which I have designed using the descriptio
written here:
http://www.har...