basic DSP with FPGA

Hi all, I'm new to FPGA stuff... i have an idea of making an audio
spectrum analyzer implemented on a fpga, displaying data on a VGA
monitor (spartan-3 starter kit). I guess I will need to filter the
audio data in some way (band pass filters for each channel?), and I
want to do it with the FPGA. That is, I'll get some ADC and plug it to
the FPGA, and then process the audio, first some basic filters for
audio manipulation (low pass, "bass boost", etc), and then move on to
something more complex..
Does anyone have some pointers of how to implement this kind of
filtering with FPGAs?

Regards,
Hernan

PS. Happy new year! (31/12/2005 9:51 PM here)

opencores.org

drg wrote:

> Hi all, I'm new to FPGA stuff... i have an idea of making an audio
> spectrum analyzer implemented on a fpga, displaying data on a VGA
> monitor (spartan-3 starter kit). I guess I will need to filter the
> audio data in some way (band pass filters for each channel?), and I
> want to do it with the FPGA. That is, I'll get some ADC and plug it to
> the FPGA, and then process the audio, first some basic filters for
> audio manipulation (low pass, "bass boost", etc), and then move on to
> something more complex..
> Does anyone have some pointers of how to implement this kind of
> filtering with FPGAs?
> 
This is basic DSP stuff.  At its most basic you're just implementing the 
math on an FPGA; in reality you're going to be doing a bunch of messing 
around with the details of the process.

You really need to know three things: DSP, FPGA design, and optimizing 
FPGA designs for DSP.  For the DSP part of it I recommend "Understanding 
Digital Signal Processing" by Rick Lyons.  I learned logic design by 
osmosis, so I can't recommend any one text.  When you get to the part of 
merging the two onto a real, live FPGA I'd check the Xilinx web site for 
app notes.

-- 

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

"Tim Wescott" <tim@seemywebsite.com> wrote in message 
news:xLSdnUIS0pfE9yrenZ2dnUVZ_sednZ2d@web-ster.com...
> drg wrote:
>
>> Hi all, I'm new to FPGA stuff... i have an idea of making an audio
>> spectrum analyzer implemented on a fpga, displaying data on a VGA
>> monitor (spartan-3 starter kit). I guess I will need to filter the
>> audio data in some way (band pass filters for each channel?), and I
>> want to do it with the FPGA. That is, I'll get some ADC and plug it to
>> the FPGA, and then process the audio, first some basic filters for
>> audio manipulation (low pass, "bass boost", etc), and then move on to
>> something more complex..
>> Does anyone have some pointers of how to implement this kind of
>> filtering with FPGAs?
>>
> This is basic DSP stuff.  At its most basic you're just implementing the 
> math on an FPGA; in reality you're going to be doing a bunch of messing 
> around with the details of the process.
>
> You really need to know three things: DSP, FPGA design, and optimizing 
> FPGA designs for DSP.  For the DSP part of it I recommend "Understanding 
> Digital Signal Processing" by Rick Lyons.  I learned logic design by 
> osmosis, so I can't recommend any one text.  When you get to the part of 
> merging the two onto a real, live FPGA I'd check the Xilinx web site for 
> app notes.
>
> -- 

Alternatively use Altera instead where the DSP tools come fitted as 
standard. You can create your own filter specification by specifying number 
of bits/filter type/number of stages/sample rate etc. The tool will then 
plot the actual achieved response which you can then tweak  dynamically. 
Code is then generated which you then drop into your design. Simplicity 
itself really if you do not want to get deep into the math.

Slurp

yes, that's my problem, the maths. I'm still in college (systems
engineering), but I have yet to learn all that advanced math to
understand even the most basic DSP text. These are all full of huge
formulas and complex numbers... I'm not trying to sound lazy but I
don't think it has to be that complicated.

I just need a routine, and to learn how to use it (I seem to have found
it at opencores
(http://www.opencores.org/projects.cgi/web/biquad/overview) along with
DSPlay to generate the coefficients. 

Regards,
hjf

drg <drgenio@gmail.com> wrote:

> Hi all, I'm new to FPGA stuff... i have an idea of making an audio
> spectrum analyzer implemented on a fpga, displaying data on a VGA

I'm not quite sure but spectrum analyzer sounds like FFT.

> want to do it with the FPGA. That is, I'll get some ADC and plug it to
> the FPGA, and then process the audio, first some basic filters for
> audio manipulation (low pass, "bass boost", etc), and then move on to
> something more complex..
> Does anyone have some pointers of how to implement this kind of
> filtering with FPGAs?

I suggest you start doing all this at a higher level, let's
say Ada or C#. You can do your researches without dealing with
FPGAs and so on.

You already know that this is DSP stuff, but it doesn't matter
if you use your normal CPU or a real DSP (probably as VHDL).

You'll see that (in general) a DSP is just a processor which
is good for MAC: multiply-and-accumulate, so to say "*" and "+",
but you can also do this with your Athlon ;)

Have a look at PCM to understand how audio data is represented
for processing, what a sample is and why PCM is almost Wave.
You can compare your results by looking at
<http://adi.thur.de/index.php?show=asound>.

Then start writing your code in a highlevel language and
use normal wave-files (or raw PCM) to feed your software
audio processor. You can write some basic effects like
"echo" by filling a buffer, dividing (lowering volume) and
adding it to the main signal after some delay (sidechain-FIFO).

You can change the volume by multiplying the signal with
a scalar.

After you've done all this, you can get a real ADC, a soft-IP DSP,
poke everything to your FPGA and use your new hardware to do
the job ;) Until then, read/write from/to wave files or
your soundcard.

I suggest you use Linux, have a look at the "sox" manpage, play
with sox -x -t raw/wav and so on. If your CPU is fast enough,
you can read vom /dev/dsp, calculate some functions and write
the results back to /dev/dsp. I usually read from wave files
and write to /dev/dsp to get an immediate impression of what
goes wrong... ;)


There is also some more audio processing software for Linux,
most of it is open source, so you can see how to implement
the FFT for your spectrum analyzer, filter effects (see LADSPA)
and so on.

I haven't ever tried a real DSP, but some datasheets claim
FFT-optimization, so it might be possible to pass a buffer
to a special asm-function, returning the FFT-transformed
of the input.

One word left: you'll probably implement your filters
in your DSP's assembler. So to say: it is all about
software and it is hardware specific. That's why you
can use a highlevel language to understand what's going
on. Filtering is more or less easy, have a look at
your opencores example:

y[n] = b10*x[n] + b11*x[n-1] + b12*x[n-2] + a11*y[n-1] + a12*y[n-2]

where x[n] represents the current input, x[n-1] is the previous
sample, y[n-1] is the previous output sample and all "a"s and "b"s
are simple coefficients for fitting the desired curve. y[n] is
your current output.

With libao, one can write this program in less than 10 minutes,
it goes like this:

procedure iir is
   b10 : constant Float := 1.2345;
   b11 : constant Float := 2.3456;
   b12 : constant Float := 3.4567;
   a11 : constant Float := 4.5678;
   a12 : constant Float := 5.6789;

   x, x1, x2, y, y1, y2 : Float := 0.0;

begin
   loop
      x := Get_Input; --  however
      y := b10 * x + b11 * x1 + b12 * x2 + a11 * y1 + a12 * y2;
      Output (y);  --  however
      y2 := y1;  --  prepare for next cycle
      y1 :=  y;
      x2 := x1;
      x1 :=  x;
   end loop;
end iir;


The convolution of the input signal with this mask isn't
that hard, all the trick is in the mask. I'm currently
doing some digital image processing with C++, in most
cases this means looping over the image, consider the
pixels in the neighbourhood and somehow adding them
to the current pixel. This is pretty much how audio
processing works: look at current, some older and perhaps
some newer samples, take them in consideration and output
something according to this consideration ;)


Get a book or google for digital signal processing.

HTH

-- 
mail: adi@thur.de  	http://adi.thur.de	PGP: v2-key via keyserver

Lieber ein Busenwunder als ein wunder Busen

Hello Herman,

Number of choices....

1....Altera  DSP Builder
2....Xilinx Coregen
3....Tyder IP Code Generator
4....Google

Regards
Bob
drg wrote:
> Hi all, I'm new to FPGA stuff... i have an idea of making an audio
> spectrum analyzer implemented on a fpga, displaying data on a VGA
> monitor (spartan-3 starter kit). I guess I will need to filter the
> audio data in some way (band pass filters for each channel?), and I
> want to do it with the FPGA. That is, I'll get some ADC and plug it to
> the FPGA, and then process the audio, first some basic filters for
> audio manipulation (low pass, "bass boost", etc), and then move on to
> something more complex..
> Does anyone have some pointers of how to implement this kind of
> filtering with FPGAs?
>
> Regards,
> Hernan
> 
> PS. Happy new year! (31/12/2005 9:51 PM here)

If it fits your terms of reference, you could also consider a 
software-only application, using (for example) the Infineon TriCore 
(TC1796) CPU. There's a development board for it (TriBoard).
This beastie runs 150MHz clock, has 12-bit fast ADC's on board, and 
hardware addressing for FFT bit-reversal, etc. Oh, and (single 
precision) hardware floating point.

stenasc@yahoo.com wrote:
> Hello Herman,
> 
> Number of choices....
> 
> 1....Altera  DSP Builder
> 2....Xilinx Coregen
> 3....Tyder IP Code Generator
> 4....Google
> 
> Regards
> Bob
> drg wrote:
> 
>>Hi all, I'm new to FPGA stuff... i have an idea of making an audio
>>spectrum analyzer implemented on a fpga, displaying data on a VGA
>>monitor (spartan-3 starter kit). I guess I will need to filter the
>>audio data in some way (band pass filters for each channel?), and I
>>want to do it with the FPGA. That is, I'll get some ADC and plug it to
>>the FPGA, and then process the audio, first some basic filters for
>>audio manipulation (low pass, "bass boost", etc), and then move on to
>>something more complex..
>>Does anyone have some pointers of how to implement this kind of
>>filtering with FPGAs?
>>
>>Regards,
>>Hernan
>>
>>PS. Happy new year! (31/12/2005 9:51 PM here)
> 
> 

drg wrote:
> yes, that's my problem, the maths. I'm still in college (systems
> engineering), but I have yet to learn all that advanced math to
> understand even the most basic DSP text. These are all full of huge
> formulas and complex numbers... I'm not trying to sound lazy but I
> don't think it has to be that complicated.
> 
> I just need a routine, and to learn how to use it (I seem to have found
> it at opencores
> (http://www.opencores.org/projects.cgi/web/biquad/overview) along with
> DSPlay to generate the coefficients. 
> 
> Regards,
> hjf
> 

Heh. The math will come in handy when (not if) you design doesn't work 
correctly. You really need to build a basic understanding of what you 
are trying to do first. How else are you going to find out what's wrong 
when things start to misbehave ? What you are trying to do sounds like a 
very interesting - but also very complex - project. I would guess that 
you need to go through the following steps - Note that you can shortcut 
the process a lot of places by using already existing modules. This of 
course will get you to your goal faster, but you will learn less in the 
process -  YMMV:

1) You need to buy/make an A/D converter module for the starter kit, 
since it does not come with one.

2) Design a filter bank. For spectrum analysis a 3rd octave bank is 
probably what you want. You can get plenty of references for how to 
design that on Google. This will involve IIR filters - so knowing the 
math (and how to implement them in fixed point) will be useful unless 
you shortcut the process and grab some already-made implementation from 
somewhere. It will REALLY help you debugging your system if you have a 
known-to-work high level implementation (Matlab, Numerical Python or C 
as a last resort) that you can compare your results to.

3) Get the filter bank implemented in some HDL (I prefer Verilog but 
others prefer VHDL). Here a book like "Digital Signal Processing With 
Field Programmable Gate Arrays" by Uwe Meyer-Baese might come in handy. 
Alternatively Mathworks and others will happily compile your high-level 
implementation directly to your FPGA to the tune of big $$$. But again - 
you will learn less in the process. If you choose to implement yourself 
I can recommend using the free Xilinx tools to do implementation and 
simulation.

4) A filter bank does not a spectrum analyzer make. You need to present 
this on the VGA output. This means building a graphics frame buffer 
structure for outputting the graphics, and then some kind of processor 
structure to fill the frame buffer in a sensible way based on the filter 
bank output. Here I guess that the best way to go would be to get an 
already made processor core like the picoblaze and use that to do the 
presentation. For the frame buffer you will be able to find examples on 
the web.

Have fun - I mean it - I for one find it to be a very interesting 
project, that you will learn a lot from if you choose to dwelve into all 
the topics involved.

-- Brian

Hi
One thing at practical side.
You should remember about simple analog Low-pass filter before A/D
converter.
If you'll forget it You will be very surprised.
It could be that you'll see signals on your spectrum analyzer that you
can't hear at all 
(Nyquist, aliasing...)

Good luck.

Jerzy Gbur