Hello!
Im looking for a simple matlab code that illustrates an interpolation
using polyphase implementation. I need this to understand the purpose with
the polyphase structure. Maybe anyone knows some sources also?
Thanks in advance
...
hi..
This is my 1st mail to this group. I am trying to do a polyphase
filter bank in dsp and thn get it into FPGA.. this is not for
decimation or interpolation. This is mainly done for radio telescope
back end in which we need 4 or 8 small channels from a big IF coming
in. Can any one help me t...
Hi,
Hoping some of you experts can shine some light on the following
regarding polyphase filtering
1...I created a polyphase FIR interpolator, sampling frequency is
upsampled by 10 from 10KHz to 100KHz. It works fine and I get the
required frequencies output on the scope. I notice however, ...
hi all,
I'm trying to build a polyphase channelizer in matlab using a prototype
filter(sqrt nyquist) as my school project. Can anyone please explain this
concept more precisely to me and what are the important things I should
consider when implementing.
Thank You.
Riddhi
...
Dear All,
Iam working subband acoustic echo cancellation.In that for
subbanding the input signal i used polyphase IIR filter.I searched in
net for polyphase IIr filter design.I got some techinical documents
related to polyphase IIr filter, I mentioned important links related to
do...
Dear All,
This is the my first mail to post our group.Iam working on subband
acoustic echo cancellation.In that for subbanding the input signal 2 path
polyphase IIR filter is used.I searched for polyphase IIR filter design in
net I got some techginical documents.In those documents one doc...
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 grou...
I have a few programs that compute and display a running waterfall
of an audio signal input to the PC sound card. The waterfall is
computed via ordinary FFTs up to 1048576 points for the highest
resolution settings.
Recently I have seen a similar program that uses polyphase FFTs
and it has a sp...
Hi,
does anyone know if there is any reference code which implements
a generic polyphase FIR resampler (up/down scaler), using Altivec
instruction set?
I searched the web (as usual), but I could not find anything
really usefull, actully I did not find anything at all.
Also non-obvious se...
Hi all,
I encounter some "problem" while using polyphase filter for
interpolation. I have implemented a 8 zero crossing with upsampling
of L=32.
The image I used are a simulated image with 4 sinc impulse response
near the four corner. All pixels values are positive.
While performing ...
I did a decimator FIR filter simply by doing the convolution only once
every M' input sample.
for example M=3:
feed x0,feed x1, feed x2, convolution, feed x3,feed x4, feed x5,
convolution, feed x6,....
Then I find many people talk about polyphase decimation filter, I wonder
if it is more effi...
in article 1132685207.779486.257470@o13g2000cwo.googlegroups.com, Anshu at
anshu.dhawan@gmail.com wrote on 11/22/2005 13:46:
> Can anyone please explain me about Interpolation using FIR filters? I
> dont understand how they can be used for interpolation.
might you understand how an FIR fil...
Praveen wrote:
> Hi
>
> I am implementing a polyphase decimation of a 8k signal by a factor of
> 20. I have designed a filter of pi/20, which means there are 20
> subbands. In case of decimation, the input gets shifted by a sample at
> each time instant and then convolved with all the...
jon8spam@yahoo.com wrote:
> Hi,
> When one designs a fractional rate change filter, when specifying
> the filter parameters do you design the filter at the interpolated
> rate or the decimated rate?
>
> For example if you have an input rate of Fs then interp by 64
> decimate by 12...
Hi,
i would like to know which is the best method for implementing a integer
sample rate converter (decimation /interpolation) in terms of quality.
the conversion rates required are (2,4,8,16) both decimation and
interpolation.i had tried with polyphase implementation and lagrange
interpolatio...
Hi All,
I am trying to debug a FM demodulation noise problem for a FPGA
prototype. The output in general looks ok, but the THD+N of a clean 60%
modulated signal is only in the mid-40's dB (noticable even by ear),
and that is mainly due to a high noise floor. The FM architecture is
using the d...
Folks,
I understand that, based on the Nobel Identities, the output of a
polyphase decimator should be identical to the output of a basic FIR
decimator. This is due to the fact that even though we decimate in the
full input bandwidth all the undersampled aliases will exactly cancel
in the fina...
Hi, All. I'm trying to use Intel's IPP polyphase resampling functions.
They are part of the signal processing set and found in the manual under
Speech Recognition Functions. I've found the manual's usage description
somewhat lacking. I don't understand it well enough to implement.
Can an...
I have a question with regards to going from a polyphase decimation
routine and adapting it to an actual filter bank to separate a large
bandwidth into smaller subbands.
I understand how polyphase filters works, that no problem, I am just
stuck on seeing how you make the transistion to a MxL (...
Dear Sir,
I am working on Acoustic Echo Cancellation. I have
implemented both AEC without subbanding and AEC with subbanding in C+
+. For subbanding of AEC I used Polyphase IIR filter(with 70dB
attenuation).
AEC without subbanding is working fine. But in the outputs
...
Let me try to summarize Farrow's idea in three bullet points:
The numbers are from the reference.
* I can use a FIR filter to interpolate one particular point between
samples, for example at an offset of 1/32 sample time
* I keep 32 different banks of FIR coefficients at hand, so that I can
c...
Hello...I created a polyphase decimating filter based on F.Harris's book
shared register design. This is a Decimation by 2, 100 tap, Fs =100MHz,
Low pass 20MHz - stop band 24MHz filter. Each MAC is 28-bit length,
14-bit ADC input, and the filter output is truncated.
The filter looks good a...
Hello,
agree. Here is one reference:
http://www.signumconcepts.com/IP_center/paper018.pdf
I've got some slides here, mostly for my own use:
http://www.elisanet.fi/mnentwig/multirate.pdf
Those aren't polished, the polyphase filter is on page 15/16.
In case of discrepancies, I'd tend to be...
Scaling the output up by a factor N, certainly improves things.
Many thanks to all in comp.dsp especially Fred for their help. Now,
next problem.....
Best Regards
Duncan
...
On 18 Oct 2005 03:39:08 -0700, asnivas223@gmail.com wrote:
> Hi,
> first of all i thank you all for your explanation, one thing i want
> to ask is that in Gardners paper he is taking only 2 samples/symbol.
> But i am having 16 samples/symbol or it may vary depending upon the
> incoming sig...
Hello...
I created a polyphase filter filter based on F.Harris's book shared
register design. This is a Decimation by 2, 100 tap, Fs =100MHz, Low pass
20MHz - stop band 24MHz filter. Each MAC is 28-bit length, 14-bit ADC
input, and the filter output is truncated.
I also created a Xilinx d...
Question for Erik, Julius, or someone else in the know:
Erik's web page says (http://www.mega-nerd.com/SRC/quality.html):
"SoX provides three methods of resampling; a linear interpolator, a
polyphase resampler and the Julius O. Smith simulated analogue filter
method."
What's the differenc...
Hi.
I am looking to add "wavetable" playback of sampled music/instruments to an
existing project. The design already uses an FPGA & has an audio CODEC
which supports playback at 48 kHz.
I want to implement pitch-shifting and mimic higher/lower notes by altering
the playback rate of the sam...
Randy Yates writes:
> julius writes:
>
> > On May 31, 8:14 am, Jerry Avins wrote:
> > > Is your filter symmetric (or antisymmetric)? If so, you can replace half
> > > the multipliers with adders (or subtracters).
> > >
> > > Jerry
> >
> > Or, use something like the Canon...
"siddharth.vaghela@gmail.com" writes:
> thank u for the reply :)
> thts wht i am doing exactly.. preparing my basics well and trying to
> get my hands on simple programs on fir filters, bit reversals, etc etc.
> i do get confused sometimes bcoz of the sheer size of material to be
> re...
coolup wrote:
> When I checked for PQMF filters in Wikipedia , I got the following
> information.
> It says that signal in odd subbands is stored frequency inverted. Does
> anybody know how the frequency is inverted?
>
>
> > From Wikipedia, the free encyclopedia
> A polyphase quad...
Hi,everybody,
I am doing arbitrary sample rate conversion. It may be very easy to do
when the up factor L and down factor M is very small(e.g. L=3, M=4). But
when L and M is very large, it seems difficult to me.
say 8-> 44.1k L=80, M=441, I have tried two method.
1. windowed-sinc method. Th...
I've been following a number of threads over the past couple of weeks
regarding fractional timing delays and interpolation, but none of them
really address my problem, so here goes....
I have a sampled received signal taken at r(kTs), Ts = sample rate and
k is integer. The sampled signal is t...
sinister wrote:
> A data analysis package I use (built on top of matlab) has a function for
> doing fractional time shifts of time series with constant sampling rates.
are you applying this non-integer time shift to a single buffer of a
signal or to adjacent (perhaps overlapping) segments of...
Hi Bhaskar,
> > I think however that the low-pass FIRs in the block diagram on page 1 are
> > still decimating by either 4 or 2 (clk/4r or clk/2r) from their input
> > rate
> > though.
>
> Well...that was the designer's choice. This chip's block diagram would not
> change if that la...