How to write a High-Pass and Low-Pass filter as soon as possible?

Dear All,

I hope to do the High/Low Pass filter for a 16bit/mono PCM audio
stream. I already study this serveral week. But I really can't
understand how it work. (Everybody tell me it is very easy.)

Could anyone tell me how to finish this job by the fastest way? I
should close my project at the end of Nov.

Emmer

Study More. Do the work.  It won't get any easier.

Dirk

dbell wrote:
> Study More. Do the work.  It won't get any easier.
> 
> Dirk

I certainly agree.  FIR filters are very easy to implement, even easier 
to design given the large number of programs out there built to 
calculate coefficients.

If you don't understand FIR filter operation then I would be likely to 
suggest that you didn't study it well enough in those several weeks. 
You should be able to learn all you need to know in about twenty minutes 
on the subject.

Emmer wrote:
> 
> Dear All,
> 
> I hope to do the High/Low Pass filter for a 16bit/mono PCM audio
> stream. I already study this serveral week. But I really can't
> understand how it work. (Everybody tell me it is very easy.)
> 
> Could anyone tell me how to finish this job by the fastest way? I
> should close my project at the end of Nov.
> 
> Emmer

We need more information on what exactly you're trying to do.  Are you
trying to low-pass filter, high-pass filter or both?

Here's a link to a digital-filter tutorial that has the added bonus of
several applets that let you design various filters to your given
specifications.  Good luck.

http://www.dsptutor.freeuk.com/

in article cZ1ff.1944$vH5.109993@news.xtra.co.nz, Bevan Weiss at
kaizen__@NOSPAM.hotmail.com wrote on 11/17/2005 23:12:

> dbell wrote:
>> Study More. Do the work.  It won't get any easier.

it depends on how much people spoon feed.  sometimes, by publishing useful
results, that's what we do.  the purpose is to save aggregate labor by not
duplicating effort.

wikipedia and google are good places to look for stuff.

> I certainly agree.  FIR filters are very easy to implement, even easier
> to design given the large number of programs out there built to
> calculate coefficients.

ya gotta get those programs.  if he/she is a student, he/she might have
access to MATLAB and the Sig. Proc. toolbox and the he/she has the necessary
programs.

> If you don't understand FIR filter operation then I would be likely to
> suggest that you didn't study it well enough in those several weeks.
> You should be able to learn all you need to know in about twenty minutes
> on the subject.

if you want to go the IIR route, you can use

    http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt .

it's pretty self-contained.  could use a few diagrams (i know Al, have
patience).

-- 

r b-j                  rbj@audioimagination.com

"Imagination is more important than knowledge."

Emmer wrote:
> I hope to do the High/Low Pass filter for a 16bit/mono PCM audio
> stream. I already study this serveral week. But I really can't
> understand how it work. (Everybody tell me it is very easy.)
>
> Could anyone tell me how to finish this job by the fastest way? I
> should close my project at the end of Nov.

Do you remember RC filters from college physics?  Hook
your PCM audio stream up to a D/A, pass the analog
signal through a resistor capacitor network, and then
redigitize.  Could you do that before the end of Nov?

IMHO. YMMV
- rhn

Thanks all. I think I should start form the basic of the DSP.
According the Jake's suggestion, I found a good web site
(http://www.bores.com/courses/intro/index.htm).
I will read first, and try to do it fast.

Ron N. wrote:
> Do you remember RC filters from college physics?  Hook
> your PCM audio stream up to a D/A, pass the analog
> signal through a resistor capacitor network, and then
> redigitize.  Could you do that before the end of Nov?

perhaps thats even faster and easyer in software, like:

double OutputSample = 0;
for (n = 0; n < NumSamples; n++)
{
    //RC-simulation
    OutputSample = (OutputSample * 90 + InputSample[n] * 10) / 100;
    //store results
    LowPassOutput[n] = (short)OutputSample;
    HighPassOutput[n] = InputSample[n] - LowPassOutput[n];
}

here cutoff depends on the out- and in-factors.
though this filter does not sound sharp, its just simple, quick and
dirty,
but probably understandable.

hope it helps, 

carsten neubauer
http://www.c14sw.de/

...I never learned about the DSP...I am not a student of E.E.
That is the reason why I can't understand the the algorithm of DSP.

If you can give me a hand, please....

Emmer

r b-j,

The problem with spoon feeding people is that often they can only
regurgitate what they were fed, and it often loses something in the
mix.  Are they really being done a favor?

Dirk