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Discussion Groups | Analog Devices DSPs | Re: DSP algorithms on FPGA/ASIC

Technical discussions related to Analog Devices DSPs (including Blackfin, TigerSHARC, SHARC and ADSP-21xx DSPs).

  

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DSP algorithms on FPGA/ASIC - Pinaki Chanda - Apr 3 15:12:00 2003



Hi all,
I am looking for some good reference on
writing optimized VHDL code for signal processing
application. Can anyone give me some reference
which talks abt smart ways to implement DSP algorithms
in ASIC/FPGA ?

Thanks and regards,
pinaki
_______________________________________________________________________
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Click here to know more!
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Re: DSP algorithms on FPGA/ASIC - Martin.J Thompson - Apr 4 7:30:00 2003

>Hi all,
> I am looking for some good reference on
>writing optimized VHDL code for signal processing
>application. Can anyone give me some reference
>which talks abt smart ways to implement DSP algorithms
>in ASIC/FPGA ? You could start by looking at Ray Andraka's website (http://www.andraka.com/)
and hang out on comp.arch.fpga (and comp.dsp if you're interested in the DSP
theory)

Cheers,
Martin --
Martin Thompson BEng(Hons) CEng MIEE
TRW Conekt
Stratford Road, Solihull, B90 4AX. UK
Tel: +44 (0)121-627-3569 -




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Re: DSP algorithms on FPGA/ASIC - Eliahu Friedmann - May 16 14:51:00 2003

Write the algorithm in Matlab and use AccelChip tool (www.accelchip.com) to move to HDL RTL (Verilog or VHDL)

It is definitely a "smart way".

Cheers,
Eli

Martin.J Thompson wrote:
Hi all,
I am looking for some good reference on
writing optimized VHDL code for signal processing
application. Can anyone give me some reference
which talks abt smart ways to implement DSP algorithms
in ASIC/FPGA ?

You could start by looking at Ray Andraka's website (http://www.andraka.com/) and hang out on
comp.arch.fpga (and comp.dsp if you're interested in the DSP theory)
Cheers,
Martin

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Re: DSP algorithms on FPGA/ASIC - Martin.J Thompson - May 19 8:49:00 2003

Eliahu Friedmann <> wrote:
> Write the algorithm in Matlab and use AccelChip tool (www.accelchip.com)to
move to HDL RTL (Verilog or VHDL)
>
>It is definitely a "smart way".
>
>Cheers,
>Eli
>
>Martin.J Thompson wrote:

I didn't actually write all that you quoted, an attribution must have got lost
there - all I said was

>> You could start by looking at Ray
>>Andraka's website (http://www.andraka.com/) and hang out on comp.arch.fpga
(and comp.dsp if you're interested in the DSP theory)

Cheers,
Martin





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Re: DSP algorithms on FPGA/ASIC - Sigmund Gudvangen - May 19 18:14:00 2003

Le Vendredi 16 Mai 2003 16:51, Eliahu Friedmann écrivit :

> Hi all,
> I am looking for some good reference on
> writing optimized VHDL code for signal processing
> application. Can anyone give me some reference
> which talks abt smart ways to implement DSP algorithms
> in ASIC/FPGA ?

Have a look at:

1) Lars Wanhammar , Dsp Integrated Circuits, Academic Press, 1999.

2) Keshab K. Parhi, VLSI Digital Signal Processing Systems : Design and=20
Implementation, Wiley-Interscience, 1999.

Regards
Sigmund.





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Need your help:FIR FILTER DESIGN using Lookup Tables - Siripon Sukuabol - May 20 3:00:00 2003

Dear All;

I need your help.
I have designed FIR Low Pass Filter using Lookup Table
and I have a problem to sum the output coefficients.
I do not know how to set up scale accumulator at final
result.

I have 20 LUT. 0-159 coefficient
after add first time and accumulate 1\2 left 10LUT
after add second times and accumulate 1\4 left 5LUT
I left 3LUT and I do not know how to scale.

I attach document file shown about diagram.

If anyone know how to solve this problem,please help
me. Thank you very much. Regards; Siripon __________________________________

Attachment (not stored)
lookuptable_fir.doc
Type: application/msword




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Re: Need your help:FIR FILTER DESIGN using Lookup Tables - Sigmund Gudvangen - May 20 11:46:00 2003

Le Mardi 20 Mai 2003 05:00, Siripon Sukuabol écrivit :
> Dear All;
>
> I need your help.
> I have designed FIR Low Pass Filter using Lookup Table
> and I have a problem to sum the output coefficients.
> I do not know how to set up scale accumulator at final
> result.
>
> I have 20 LUT. 0-159 coefficient
> after add first time and accumulate 1\2 left 10LUT
> after add second times and accumulate 1\4 left 5LUT
> I left 3LUT and I do not know how to scale. You are presumably using fixed-point arithmetic, so the sum will grow by one
bit (worse case) for each summation. Hence you need to scale, i.e.
right-shifting the word by one bit per addition. The scaling can either be
distributed (i.e. you shift by one bit after each addition) or you may use
wide enough register to accommodate the word-growth and then carry out a
multi-bit shift of the final sum. The latter option leads to lower round-off
noise, but at the expense of increased area requirements (I presume we are
talking about an ASIC or FPGA). However, the best option might be to simply
pre-scale the filter coefficient vector. That way you don't need shifters.

Regards
Sigmund.




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