Hello all,
Following a small research I conducted after posting my question about
antialiasing of ADC samples to sci.electronics.design, I wrote a small
and simple article to explain to myself, once and for all, the deal
about analog & digital antialias filters, multirate systems and how
thes...
Hello,
How to implement Multirate filtering on a DSP. My problem is how to
have a flow in my code, How should the loop of the code be. This
because my data is coming at a higher rate (4 MHz) and i am
downsampling to slower rate (i.e 500 kHz). so there is no continous
flow.
Any suggestion or ...
Dear all,
I was thinking of purchasing "Multirate Signal Processing for
Communication Systems" by F.J.Harris. Has anyone here used this book,
and if so, what are your opinions?
Thanks in advance,
--
Oli
...
I have 2 serially connected multirate filters. One has 18 taps and the
other has 61 taps. The decimation factors are 2, 2. Now I want to combine
the 2 filters into one. The new coefficient file has 78 taps by using
convolution. In FDAtool, the order is 137 after cascading. I think it is
from 137=17+...
Version 3.1 of Tyd-IP Code Generator has now been released. Now
supporting multirate filter design, designers can create a wide range
of IP, including filters, FFTs, NCOs etc. which can be targetted at
any FPGA or ASIC. Until, the end of May 2007, the multirate
coefficient and VHDL module, will ...
Hi,
I have to design a 6th order IIR BPF that will allow a multirate filter
structure with D==3.How do i start?Is there a matlab function that does
this,I mean also incorporates the multirate structure with design of an IIR
BPF like butter().I wil greatly appreciate any guidance on this.
...
"seb" wrote in message
news:23925133.0401131910.7f22e0a2@posting.google.com...
> Hello,
>
> i am looking for decimation and interpolation technique in order to,
> given a sampling rate fs, obtain a new sampling rate like (a/b)*fs.
>
> A way to to do is to decimate and then use line...
Please refer to
www.intersil.com/data/fn/fn3651.pdf
page 10 top and page 9 bottom.
Please correct me if I assume something wrong.
On figure 12 (pg 10) you see the aliasing profile of the CIC and how
it does it translate to baseband. The filters performance can
therefore be estimated. Quit...
ssnyder wrote:
> I am trying to design a set of bandpass filters for an audio spectrum
> analyzer. Since human perception of pitch is spaced logarithmically
> according to frequency, my low pitches have quite low and narrow bandpass
> frequencies.
This is a very good application for mu...
Hello,
Working on the design of decimation filter for multistage, sample rate
conversion.
I have designed a filter which represents G(z). With freqz(b,1,1024) I can
plot the frequency response of the filter (the coefficients are in "b").
However, its output will be downsampled by, say M = 1...
Hi,
I'm trying to implements a pitch shift effect (up / down a .wav sound
44100Hz 16bit to different semitones in the range of around 1 octave).
My problem is that I need to do it on about 15 sample simultaneously in
real time.
The classic pitch shift algorithms use too much of processor time...
hi folks,
I need sum help regarding multirate systems.
I am currently working on a project where we r developing a Transfer
Function model for Kalman Filter (Actually EKF). We have developed a
Kalman Filter for Avionics System of an autopilot.
In this filter we have GPS (measurements for KF) ...
Hye,
I have a question to try to find a way to improve my processing.
I use a multirate filter with decimation.
There is a decimation by 40. One FIr is used then decimation, and the 1
IIR with decimation, then a second IIR with decimation, then again oner
IIR and decimation and at last one F...
It is stated that in case of narrow transition band filters, multirate
filtering provides us with a computational efficiency as compare to
standard time invariant filters. The idea is to reduce the sampling rate
(less number of samples) and to use simple, low order filters (less number
of operations...
i am designing one 1/3 octave band spectrogram analysis
I use multirate filter bank to realize that spectrogram. It goes this
way
1)
first, from the biggest frequency value, I use three IIR bandpass
filter, then
calculate the std value.
2)
Then decimate the input by 2 through(one 30 order...
> 1. Go to the library
> 2. Get a classic book on multirate processing by Rabiner
> 3. Don't ask any more stupid questions
4. Close all newsgroups.
...
On Feb 15, 11:37 am, lakshmanan.meyyap...@gmail.com wrote:
> I am trying to develop a data anomaly detector. I basically want to
> detect clipped data, spikes, and drifting data to begin with. Any
> suggestions on how to do it.
What do you mean by drifting?
If you're trying to detect hig...
Hello,
I wrote a blog article a while ago, maybe it's useful (or maybe not...)
http://www.dsprelated.com/showarticle/22.php
The filter is designed at the intermediate high rate.
Cheers
Markus
...
"fjwoemcu" wrote in message
news:6pOdnQMkVc9vLzrYnZ2dnUVZ_vShnZ2d@giganews.com...
>
> In an ADC - rate converter/filter - DSP chain,
> the interpolator(cubic, linear...) may be placed between sigma-delta ADC
> and rate converter/filter to approximate 310MHz-sampled data to
> 300MHz-...
I'm working my way through fred harris's Multirate Signal Processing text
and have found at least one error. Does anyone know of a site where errata
have been published for this book? Maybe it's time to start an unofficial
site. I'll make the first contribution!
Hugh
...
Multirate digital signal processing by N. J. Fliege
Wavelets and filter banks by Gilbert Strang
Digital Signal processing, system analysis and design by Paulo Diniz
...
Aaron wrote:
> > There is no silver bullet. You can do Hilbert transform using multirate
> > filterbanks; that takes less computation.
>
>
> That's useful information but as Al pointed out it won't really solve
> my problem as I still require the phase shift close to DC and multira...
On Wed, 14 Dec 2005 16:44:44 GMT, Vladimir Vassilevsky
wrote:
> > ... assuming that it was designed as an approximation to a Gaussian
> > response.
>
> No. Assumming the reasonable rolloff speed and passband flatness.
You mentioned exp(-x^2) specifically. That's a Gaussian.
> > ...
Refer to: J.J.Shynk, Frequency Domain and Multirate Adaptive Filtering,
1992, Signal Processing Magazine.
The performance of the algorithms is usually evaluated using white
noise and the stepsize should be taken into account also.
Leans Nelson
...
DSP Gurus:
I'm familiar with the classic interpolate-filter-decimate approach to
multirate DSP. I've heard that the chirp-z transform can be used to
resample a signal also. Can anyone provide additional information or
references (other than Google search results)?
Thanks,
OUP
...
The proof the spectrum for decimated signal is given elegantly in the
reference below :
Multirate systems & filter banks : P.P. Vaidyanathan (ISBN 0-13-605718-7)
Pages :103-105
Regards,
Yogesh P G
Bangalore , India
...
Hi guys,
I am looking for some summer school/courses in Euro or USA 2007 regarding:
multirate, subband signal processing
OR
beamforming, microphone array
But it is really hard to find those summer school/courses. Does anybody have
any clue? Your help is really appreciated!
Best regar...
If I'm following you correctly, you want to use the output of the polyphase
legs as the input to the FFT (ie with D legs, you use a D-pt FFT). Don't
FFT an individual leg, and don't add the legs and FFT that. I could be
wrong however. Check out Fred Harris's multirate DSP book for more info.
...
Hello Stephane, here are some more category suggestions
- Theory -> Coding and Information Theory
- Theory -> Multirate Signal Processing (Interpolation and Decimation)
- Theory -> Wavelets and Filterbanks
- Theory -> Simulation/Implementation in Fixed Point
- Theory -> Modulation
- The...
Hi,
In interp.c - interp(), the while loop decrements the input sample counter
first.
while (--num_inp > 0) {
So won't it miss the last input sample? Not that you would ever call interp()
one sample at a time, but if you did, it would do nothing. --num_inp would
make the 1 a 0 and th...
Michael,
"Multirate Systems and Filter Banks" by P.P. Vaidyanathan is a
wonderful reference.
Also, "Wavelets and Subband Coding" by Vetterli and Kovacevic is
also ok.
hth,
cf
"M. Wirtzfeld" wrote:
>
> Hello,
>
> Does anyone have good references on the topic of filter-banks...
Rune Allnor wrote:
>
> Nevertheless, most DSP books I know of on the primer/intro/intermediate
> levels do include a discussion of these issues.
My favorite, "Digital Signal Processing" by Proakis and
Manolokis does. I've rarely been let down by it when I want
to understand somethin...
A filter H(z^m) before the downsampling can be implemented alternatively as
H(z) after the downsampling. That may be sort of trivial, still it's in
multrate textbooks.
A general H(z) filter before the downsampling cannot be implemented after
the downsampler in the general case: The former might b...
hello,
be aware, I'm new to DSP algorithms :)
I have more pcm streams. first I have to change the sample rates of
some pcm streams and than I have to "add" all streams in a (new) pcm
stream (with a given sample rate).
How can I do this?
As I read on the dspguru.com, changing the sample ...
Andrew Xiang wrote:
> Does any have an efficient implementation of polyphase filterbank in C? M=32
>
There's a not-too-shabby one at http://www.dspguru.com in the FAQ
section (under Multirate filters).
--
Jim Thomas Principal Applications Engineer Bittware, Inc
jthomas@bit...
Hi,
Is it so that "high Q" in your requirement translates into "steep edges"?
If so, some thoughts:
Under some circumstances it may be better to
-downconvert the signal in the digital domain to 0 Hz,
-use complex *lowpass* filters
-and translate back up to the center frequency.
Maybe...
This question relates to the Multirate FAQ at the dspGuru site:
http://dspguru.com/info/faqs/multrate/interp.htm
I have a 256 element array that I want to interpolate at a variable
density, between 4 and 10.
Let's take the first case, i.e. the interpolation factor or density, L
= 4.
Thus...
Hello Earthlings,
I recently ran across some material in
Vaidyanathan's "Multirate Systems and Filter Banks"
DSP book that discusses a way to improve the
computational efficiency of polyphase filters
used in non-integer decimation applications.
Vaidyanathan's description (starting...
> Y(e^jw) = 0.5 [X(e^jw/2) + X(e^j(pi+w/2))]. (A)
> I have a problem with your equation (A) above. I can't see straight away
> why you get two terms there.
Have a look at Gilbert and Strang "wavelets and filterbanks" bottom of
page 91 or Vaidyanathan "multirate systems and filterbanks" t...