# Need help in Designing Filter using DSP

Discussion in 'Homework Help' started by kellynyp, Mar 31, 2009.

1. ### kellynyp Thread Starter New Member

Mar 31, 2009
8
0
Hi,

I new in using a software Analog Devices VisualDSP++ (ADSP-BF561) to develop a program on FIR Filters... While doing my program i encounter a problem on the Coefficient part cos i have no idea how to get the coefficient values...

Thanks alot

Last edited: Apr 1, 2009
2. ### thatoneguy AAC Fanatic!

Feb 19, 2009
6,357
718
That may be better answered on the AD DSP Code forum, though some members here may be able to help.

Which coefficient part?

3. ### kellynyp Thread Starter New Member

Mar 31, 2009
8
0
Hi thatoneguy,

I not sure how they implement the coefficient values... I went to this website --> http://www-users.cs.york.ac.uk/~fisher/mkfilter

Enter this
================================================
Sample rate: 8000000
Corner frequency: 1000000
Enter the value of ``beta'', in the range 0 to 1: 0.3
Impulse length: 33
Raised-cosine response: yes
Compensation function: yes
Truncate coefficients to 12 bits
Lower limit (dB), or blank for linear scale: -80
================================================
Submit Query
================================================

than i get the result...

this is the part that i not very sure how to get those values...

=======================================================
static float xcoeffs[] =
{ +0.0004882812, -0.0136718750, -0.0268554688, -0.0253906250,
-0.0009765625, +0.0405273438, +0.0732421875, +0.0654296875,
+0.0014648438, -0.0981445312, -0.1772460938, -0.1621093750,
-0.0058593750, +0.2812500000, +0.6201171875, +0.8945312500,
+0.9995117188, +0.8945312500, +0.6201171875, +0.2812500000,
-0.0058593750, -0.1621093750, -0.1772460938, -0.0981445312,
+0.0014648438, +0.0654296875, +0.0732421875, +0.0405273438,
-0.0009765625, -0.0253906250, -0.0268554688, -0.0136718750,
+0.0004882812,
};
=======================================================

i need help in writing a problem on filter which include this coefficient part...

pls help me in anyway you can...

thanks alot

4. ### thatoneguy AAC Fanatic!

Feb 19, 2009
6,357
718
The page generates a table of precalculated cosine function values to the specified bit accuracy, this saves a lot of processing time on the DSP.

The source code for the program is here, it is written in awk. awk is a language sort of like C, but scripted. It is available on most *nix systems such as FreeBSD and Linux. There is an awk port for windows as well.

mkfilter source: http://people.freebsd.org/~ariff/z/mkfilter.awk.html

awk for Win32 (winxp+) is at: http://gnuwin32.sourceforge.net/packages/gawk.htm

5. ### t_n_k AAC Fanatic!

Mar 6, 2009
5,448
783
Hi,

You have to specify and design the FIR filter - there are software tools that do this for you and usually have an option to export the optimized filter coefficients to a data file.

I don't know anything about Analog Devices VisualDSP++ (ADSP-BF561) - doesn't it include a FIR filter design utility?

In any case you should start with a filter specification.....

Is it low pass, high pass, band pass?
What are the pass band and stop band frequency limits.
What is the sampling frequency?
What is the stop band attenuation?
What is the pass band ripple?

And so on ....

Scilab (free) has a FIR filter design tool.

6. ### t_n_k AAC Fanatic!

Mar 6, 2009
5,448
783
Sorry. I didn't realize the same thread was active in Programmer's Corner.

7. ### kellynyp Thread Starter New Member

Mar 31, 2009
8
0
LoL, its ok

For the specifications, i was given
- Frequency: 385 to 485 Mhz
- Signal Bandwidth: 10Mhz
- Sampling Rate 1.5Gsps

if im not wrong, it includes low pass, high pass, band pass and band stop.

Last edited: Apr 2, 2009
8. ### kellynyp Thread Starter New Member

Mar 31, 2009
8
0
To thatoneguy,

thanks for the links, it helps me alot...