Basic Filter Implementation

Started by March 4, 2007
I'm a beginner when it comes to DSP but I need to learn how to
implement basic filters on a DSP board. I'm using an ADSP-21369 EZ-Kit
from Analog Devices. I understand how to generate coefficients using
Matlab but how to I use these coefficients to filter my input signal.
Can someone explain using either code or just an explanation. Any help
will be greatly appreciated.
Dan wrote:
> I'm a beginner when it comes to DSP but I need to learn how to
> implement basic filters on a DSP board. I'm using an ADSP-21369 EZ-Kit
> from Analog Devices. I understand how to generate coefficients using
> Matlab but how to I use these coefficients to filter my input signal.
> Can someone explain using either code or just an explanation. Any help
> will be greatly appreciated.

Can't demonstrate with code as I've never used this processor.
I'll assume you're talking about implementing a plain FIR filter
without anything fancy like decimation or interpolation, half
banded filters etc..

Implementation is quite simple.

1- Calculate filter impulse response and store the resulting
N coefficients in a table. It can help if this table is
circular, but it's not strictly necessary if the processor
(some do).

2 - For each signal channel to be filtered you need this state
which must persist between samples (filter invocations)
A - A circular buffer of N previous samples (typically initialised
to all zeros).
B - A pointer to this buffer, initialised to start of the buffer.

Then for each input sample you..
1 - Write the sample to the circular buffer and post increment
the buffer pointer (using circular addressing). The pointer
will now be pointing to the *oldest* sample in the buffer.

2 - Initialise a pointer to the start of the N coefficient table.

3 - Initialise an accumulator to 0.

4 - Repeat N times
Read sample from circular buffer and post increment circular
Read coefficient from coeficient table and post increment
coefficient pointer.
Multiply sample by coefficient and add to accumulator

5 - The accumulator will now contain the filtered output
sample. Also, note that circular buffer pointer has been
incremented (N+1) times in total. (I.E. it has been advanced
1 place (N+1 MOD N = 1), so again points to the oldest sample
which will be overwritten by the next input sample.
If you're using circular addressing for the coefficients then
this pointer has been advanced N times so should point to
the start of the table again (does not need re-initialising
if you're processing several input samples in 1 batch).

6 - If this is the last input sample in the current batch then
save the current circular buffer pointer.

Note:
1- Strictly what I've just described performs a correlation, not
a convolution (impulse response will be coefficient table
reversed). For symetric FIR's (most are) this will not matter.
If the desired impulse response is asymetric then you need
to reverse the coefficient table.
2- Lots of detail about word lengths, scaling, saturation, rounding
and fractional arithmetic ommited.
3- You should be able to get the loop described in 4 down to
1 instruction per coef (filter tap). Typical DSP pseudo
code would look like this..

Repeat N-1 times; // Not N!!
accumcum+sample*coef, sample=buffPtr++, coef=coefPtr++; // 1 instr!
accumcum+sample*coef; // Final step

HTH

Regards
--
On Sun, 4 Mar 2007, Dan wrote:

> I'm a beginner when it comes to DSP but I need to learn how to
> implement basic filters on a DSP board. I'm using an ADSP-21369 EZ-Kit
> from Analog Devices. I understand how to generate coefficients using
> Matlab but how to I use these coefficients to filter my input signal.
> Can someone explain using either code or just an explanation. Any help
> will be greatly appreciated.

If it's an FIR you can use a standard subroutine with one vector coming
from Matlab and one vector coming from your input data. Use circular
buffers and change the input data pointer but not the filter data pointer.
The standard subroutine should do the multiply/sum for you.

The main trick is to ensure it's a circular buffer. Sometimes that just
easier to do in assembler - but I'm sure there's others here who can
explain all the #pragma's it takes to get circular to work in C. With a
circular buffer, you just add one data value, change the starting point of
the buffer, and go.

To input the data file, either use a #include or set up the
.var filt_coef="filename"; form in assembler. Lots of people here can
fill in details (I usually use assembler and brute force, so I'm not much
help :-)

Patience, persistence, truth,
Dr. mike
Dan wrote:
>
> I'm a beginner when it comes to DSP but I need to learn how to
> implement basic filters on a DSP board. I'm using an ADSP-21369 EZ-Kit
> from Analog Devices. I understand how to generate coefficients using
> Matlab but how to I use these coefficients to filter my input signal.
> Can someone explain using either code or just an explanation. Any help
> will be greatly appreciated.