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Hello newsgroup,
the book "Guide to Signal Processing" shows a recursive filter, the
chebyshev filter, implemented either as LowPass or as HighPass version.
How can you use a Chebyshev in a singel stage as a BandPass filter?
A BandPass implemented as FIR is formed by adding the Low- and H...
2007-05-16 11:38:00
Hi
This is regarding usage of chebyshev polynomials for function
approximation. I work on this problem during my spare time.
The Chebyshev polynomial coefficients are obtained from unequally
spaced zeroes of the function. But, if we use the same Chebyshev
polynomial to approximate a periodic ...
Hi all,
I need the above, but I can't find it in any of the books I have
handy and every google search I try turns up nothing but references
to Matlab :-(.
The Chebyshev I analogue prototype is:
H(s) = 1 / (s^2 + s / Q + 1)
and I was thinking of subtracting the above from 1 but I'm...
jungledmnc - 2008-07-01 06:57:00
Btw. I'm pretty newbie in this, what is exactly the bilinear transform in
this case? I've read about the pole-zero method, but this does not seem too
close. Just please give me a hint :-).
And one more not very celver question : What is exactly the difference
between filters like Chebyshev, biqua...
JM1970 - 2009-09-03 22:54:00
Hello,
I am trying to model the effect of group delay of an analog filter with
complex transfer function H(f) on a phase modulated signal s(t), with a
complex spectrum S(f). The analog filter I would like employ is a Chebyshev
n=4, with 0.1dB ripple, but the exact filter is not important right now...
dacrow wrote:
> I'm trying to build a Low Pass Reconstruction Filter for use after a DAC .
> The thing is it has to be a Zi = 300 Ohm input and the fc has to be 37 MHz
> and 50 dB attuenuation @ 118 Mhz and the passband ripple has to be < 1dB.
> Already tried the Butterworth but the coils...
2005-10-05 13:59:00
Hi
What polynomial are you using for approximation? You are better off
using the Chebyshev polynomial which has the minmax property (minimizes
the maximum error). But of course the problem is that the Chebyshev
polynomial is based on non-uniform sampling points and if your samples
are uniforml...
2007-07-18 11:09:00
Hi,
I would like to know on the implementation complexity issues beween
FIR Filter vs. IIR Filter (Chebyshev).
I know that IIR is more complex than FIR. But, in case of the FIR
filter has higher order number than IIR, FIR can be more complex.
If I have IIR Chebyshev 10th order vs. FIR with ...
dpenev@yahoo.com (Dimitar Penev) writes:
> In Matlab there is Hanning function which implement cos^2(t) window.
> It has -18 db/oct roll off. I need to estimate the signal PSD using
> window with higer roll off. The sin^4(t) seems adequate for my
> purpose but it seems it is not implemente...
Scott Hemphill - 2009-11-20 16:50:00
"kc6zut" writes:
> I need an algorithm for computing the exponential of a real number using
> only elementary operations (addition, subtraction, multiplication and/or
> division). I have a PLC (Programmable Logic Controller - used in
> industrial controls) as the processor. It has no ...
Scott Gravenhorst - 2008-09-23 13:35:00
What are the common method(s) for preventing aliasing of arbitrary
waveforms generated within a DSP application? I understand that
waveforms supplied as analog signals and presented to an ADC must
first be lowpass filtered to remove harmonics above the Nyquist
frequency, but what about generate...
On 5 Jan, 04:58, "Steven G. Johnson" wrote:
> Also, it's not at all clear to me how
> you think templates help you gain performance in the above code vs. a
> C-style explicit implementation
The rationale for the question is that templates displays
all the code to the compiler at once. I...
On Feb 5, 11:44=A0am, Rune Allnor wrote:
> On 5 Feb, 17:28, "superraton" wrote:
>
> > I have a problem when I calculate the coeffs of my chebyshev IIR filter=
s
> > (via Bilineal Transformation), and I hope someone can help me!
> > The thing is that when I compare my coeffs with ...
Triff - 2005-02-09 19:55:00
I'm not quite sure what I'm expecting as an output!
I seem to be getting a square wave from a sine wave with an continually
decreasing frequency!
Is this right?
Cheers
Triff
...
Complex Chebyshev Approximation for FIR Digital Filter Design
Lina J. Karam & James H. McClellan
if any bidy has this paper please email it at
danishzzz@yahoo.com
thank u
...
Some bandpass filters I have heard of:
1. Chebyshev
2. Butterworth
3. Raised Cosine
What are the attribtues of theses filters that make ones more useful
than another in a given sitation?
Iasac
...
julius - 2007-05-24 10:33:00
On May 23, 7:11 am, "c1910" wrote:
> hi,
> i'm very new with DSP FIR.
> i need to make a Bandpass filter with FIR design. first, i use the AM
> signal for the input, then i want to get the information signal by
> filtering the AM signal.
> The problem is to get the value of transfer f...
sasidhar - 2008-11-21 11:42:00
Hi all,
I am trying to calculate derivatives of chebyshev coefficients using FFTW
for cosine(x). However I am not getting back the right values when I
perform inverse chebyshev transform. Can anyone check the code below and
tell me the possible errors. Thank you.
program test
im...
A million of times of thank you, to the guys who have helped me in
this problem. Before I saw your suggestions I was trying to compensate
for that poor droop by using a pre-warped 2nd order chebyshev
(implemented on dsp and not on FPGA). I must say that it also gave
quite some good performance ...
In article bw9zb.50644$jf4.2789643@news000.worldonline.dk, Jack L. at
jack_nospam@nospam.dk wrote on 12/02/2003 18:47:
> Quote from Simon Haykin's book "Adaptive Filter Theory", 4th edition, on the
> Wiener filter (chap. 2):
>
> "We now summarize the essence of the filtering problem with...
On 29 Jul, 20:13, Jerry Avins wrote:
> William Hughes wrote:
> > ... why give up the advantages of a high level language when
> > you do not have to? Yes there are times when a good assembly
> > programmer can beat a good compiler in terms of speed, and situations
> > where it m...
Joerg - 2006-03-08 19:51:00
Hello Rick,
> Engineers shouldn't have a tough time
> gettings dates. I say that because:
>
> "Filter designers get a better response."
>
Interesting :-)
But Chebychev wasn't exactly Casanova, although rumors have it that he
had a daughter and in public he never admi...
mnentwig - 2007-09-12 06:25:00
Well...
I've got a program on my web page to find an inverse filter, but it will
give garbage on an "impossible" problem - such as inverting an ideal
highpass.
Anyway, feel free to experiment.
http://www.elisanet.fi/mnentwig/webroot/nonminphase_inverse/index.html
Note that it makes extens...
I'd really like not to reinvent the wheel for my current hobby
project, so I beg of all DSP people out there - what are the standard
filters used for 16X interpolation/decimation? Are they windowed-sinc,
Chebyshev, Butterworth, or something else? And what are the
coefficients, can I find tables ...
2006-07-26 00:26:00
Hello all,
i am designing a low pass filter and high pass filter to pass my
audio samples. in matlab i checked out both butterworth and
chebyshev(cheby2) 3rd order filter.
I have observed there is a lot of differance in the phase plot of
both the filters.
Now my query went to what ...
The classic IIR filters have either one or none of the real poles. However
when designing an analog part, it is often possible to stick many additional
1-st order RCs into the schematics. Those realpole RCs can improve the
performance without any side effects.
This seems to be a very typical s...
Jerry mentioned the Audio EQ Cookbook at harmony-central.com but there are
problems with answering your question directly.
in article m5mdnTcdJ5_tMrjfRVn-jw@giganews.com, krish at
career4krish@rediffmail.com wrote on 03/02/2005 07:20:
> Topic : To design an IIR HPF filter using IPP(Intel In...![Re: [Question] How to get BW or Q from slope(db/oct) for HPF](http://cdn.dsprelated.com/images/icon_more.jpg)
SunLei - 2007-01-01 08:18:00
hi,
On the FPGA implementation of decimation filters, which type of decimate
filters do you prefer?
I am designing a IF DDC converter, and wana use FPGA to implement the
decimate filter, but I found the Half-band lowpass and the euipripple fir
filters can hardly stop the out-band...
Is anyone familiar with the following paper?
Preuss, K, "On the Design of FIR Filters by Complex Chebyshev
Approximation," IEEE Trans. Acoust., Speach, Signal Processing, vol.
37, pp. 702-712, 1989.
The author uses a notation I am not familiar with. Specifically, see
equation (8):
| E(Om...
itsh11 - 2008-06-26 13:30:00
I am trying to design a second order digital IIR band stop (notch) filter
with the following specs:
3dB cut off frequencies: 55Hz and 65Hz
I want the notch at 60Hz with atleast 90dB attenuation at the 60Hz.
Sampling frequency: 200hz
I tried various filter configurations like a Butterworth or...
"Rune Allnor" wrote in message
news:f56893ae.0408310548.8ba60b4@posting.google.com...
> Hi all.
>
> I'm playing a bit with window functions in filter design etc, and
> have come across a couple of gruelling expressions. More specifically,
> the Chebychev window [1, eq. 5-17], the Tuk...
Fred Marshall wrote:
> So: why should an equiripple all pass have a unit sample
> response that approximates or *is* a sampled Bessel function of
> some order?
Not sure if it leads anywhere, but the Bessel Fourier transform is
F{J_n}(w) = { 0, |w| > 1; 2*j^n*T_n(w)/sqrt(1-w^2), |w| ...
qhash - 2007-06-27 09:07:00
Hi
I am having the problem with the amplitude correction also. I make
measurements in the freq. domain ,then I am going to the time domain and
cutting off the reflections. After this I am going back to freq. domain.
When applying the gain coherence coefficients from Harris paper, my
windowed sig...
Greg Berchin wrote:
> > 1. =C2uild a biquad HPF with Fc =3D 20Hz and Fs =3D 48kHz.
> > 2. Apply a small signal to the input.
> > 3. Observe the amount of trash at the output.
> =20
> =20
> 20Hz? I thought that we were talking about 100Hz.
I suggested to try lower frequency just to m...
Alex - 2009-08-04 18:58:00
Hi all,
I'm new in this newsgroup and I thank you in advance for your help.
My problem is the following. I have a couple of signals (X, Y) and I
want to separate the two sources that compose it (X1,Y1) and (X2,Y2).
I know that (X1,Y1) has low frequencies from 0 to ~3500Hz and that
(X2,Y...
On 30 Jul 2003 05:02:11 -0700, jomfrusti@image.dk
(=?ISO-8859-1?Q?Ren=E9?=) wrote:
> Is it possible to get a fast fix point Arctan routine written in C
> using the principle with a lookup table or perhaps another principle?
>
> Regards,
> René
Hi,
Look-up table methods are the fas...
On Oct 21, 4:41 am, wabehtd...@alumni.com wrote:
> On Oct 21, 5:08 am, Tim Wescott wrote:
>
>
>
> > On Sat, 20 Oct 2007 08:36:11 -0700, wabehtdieh wrote:
...
> > > If you put a sine wave in
> > > do you get a sine wave out?(Linear)
>
> > Nuh uh. There's a long thread on ...
Hi group!
I'm an engineering student and I'm trying to implement an IIR filter.
It's a 6th order Chebyshev LP filter.
We perform L2 norm scaling of the coefficients.
To simulate the implementation environment we programmed the whole thing
in Matlab, rounding the products in the appropri...
bharat pathak - 2008-02-21 11:16:00
Hello All,
I want to understand how spectral leakage changes the
phase spectrum of a sine wave?
Also when I apply window (blackman harris 11 Term or
chebyshev window with 300db attenuation), I see a
significant amount of reduction in spectral leakage
in ...
jajo - 2006-10-19 10:13:00
Hi,
I am modelling the physical layer of an 802.11 transmitter (to
implement it into an FPGA). At this moment I want to design the digital
filter responsible of pulse shaping (before the DAC). My doubts are:
1. FIR or IIR filter?, why?.
2. For each option (FIR or IIR) there are different ty...