>Aside from the obvious of an accumulator being a low pass filter, we
>normally hard code the time consuming parts of the digital filters,
>presuming that your filter coefficients are not going to change. We either
>pre-do the multiply of factors based on a trig function times a constant or
>tweak the multiply. The former is done by doing the stage multiplication
>off line and interpolating results of table look-ups. The latter is done by
>skipping the bit tests on the multiplicand and straight-line coding the
>shifts.
>
>Obviously, the time per cycle of multiply and accumulates, as with
>everything in DSP, is dependent upon your sample frequency. For mains
>power, ya got a lotta time. For voice, you'd better be doing some real
>tricky stuff in a PIC if you're doing the number of orders and resolution
>that you are looking at. We have had great luck by varying the sample
>frequency and type of filter until you get to coefficients that either
>disappear or are very easy multiplies. You can do this type of iteration
>easily with filter design packages, although normally the output is in
>terms of C code or asm for the popular DSP's.
>
>Tom
>
>On Thursday, March 05, 1998 2:22 PM, Craig Webb [SMTP:
.....lucidKILLspam@spam@MAGNET.CA]
>wrote:
>> Thanks everyone for the WWVB (Boulder or Fort Collins?) info on the 60kHz
>> time standard.
>>
>> I am also interested in finding out if anyone has done digital low pass
>> filtering. I need fast code that can do about 4th order (and ideally up
>to
>> 6th order) filtering on 13-bit signals. Does anyone have this or know
>where
>> I can find it?
>>
>> Thanks.
>>
>> C. Webb
>
>