Reply posted by Mikael Holm on December 11, 2001
I'd design FIR-filter for that particular box.

Miffe

Reply posted by Phill Graham on December 11, 2001
Hey Miffe,

While this is a fine suggestion, it's not very practical for live work. The time involved for the FIR filter is approximately the same as the length of sample to resolve that frequency with a fourier transform. This is a substantial period!

Posted by Mikael Holm on December 11, 2001
I'm dealing with a subprocessor here. 35 seconds propagation delay ;-)

During my first year in university, I was told I should use recursive algorithms whenever it's possible because it's more powerful way to do it. Well, this recursive filter just kills me. Longest sample I run through taps is ~48ms.

Miffe

Reply posted by Mikael Holm on December 10, 2001
With FIR filters, you can make EQ filters with close to zero phase shift. Ideal filters are also possible but then you would have generated an oscillator ;-)

Miffe

Reply posted by Mike G. on December 11, 2001
Where can I find out more about FIR filters?

Mike G.

Posted by Andy Peters on December 11, 2001
I can't think of anything "simple." All of the books I have assume some math (calculus) background. There's no book, a-la "Digital Filters for Dummies."

However, Digital Filters
http://www1.fatbrain.com/asp/bookinfo/bookinfo.asp?theisbn=048665088X&vm=
by R. W. Hamming, is excellent. This is a small book, and it's only $11. However, it's eminently readable. Hamming eschews mathematic rigor (and he's got a funny comment about that: "mathematical rigor...all too often leads to rigor mortis.") for explanations that make sense.

[As you read about digital signal processing, you'll come across the important concept of windowing. There's a handful of "standard" windows; one is the Hamming window.]

In addition to Proakis, the other canonical DSP book is Digital Signal Processing
http://www1.fatbrain.com/asp/bookinfo/bookinfo.asp?theisbn=0132146355&vm=
by Oppenheim and Schafer. It's expensive ($100) but covers the whole subject in clear detail.

I also have an excellent book at work that's got exercises in DSP with Matlab, but I can't seem to find it on Fatbrain.

This may be interesting.
http://www1.fatbrain.com/asp/bookinfo/bookinfo.asp?theisbn=0130909998&vm=

andy

Reply posted by Phill Graham on December 11, 2001 at 19:29:48:
Andy,

While I was interning at AMD, I started on a book called "DSP using Matlab" in the AMD reference library. I didn't come close to finishing it, but it looked like it was going a promising direction.

Phill Graham

Reply posted by Mikael Holm on December 11, 2001
Chapters 14 through 21 (Introduction to Digital Filters...Filter Comparison)
http://www.analog.com/technology/dsp/training/materials/dsp_book_index.html
touch the art of making different digital filters.

Also "Digital Signal Processing" by Proakis and Manolakis gets a recommendation from me. Andy Peters has some other books in his mind also.

Miffe

Reply posted by Tom Young on December 10, 2001
But why is it we see virtually no FIR processors ? (This is not bait)

Tom Young
Electroacoustic Design Services

Reply posted by Andy Peters on December 11, 2001
Good question.

Too much delay at low frequencies to be useful in real-time audio systems?

Reply posted by Mikael Holm on December 10, 2001
Probably because they are more or less purpose designed for special systems. HK Audio and German Audio Engineering first come to my mind. If you have read the DFC paper by HK Audio you'd know that after each, even slight, adjustment you have to compile the filter again causing few seconds delay before new filter can be loaded.

It's not impossible but to correct phase you have to know extremely well the signal chain after controller. For example only few manufacturers give the phase response specs for their amplifiers. Usually those are within' +/-25% but in some cases, like LAB4000, it's +/-5% if i remember it right.

Miffe

Reply posted by Ron Riedel on December 11, 2001
FIR filters have phase shift. However, if they are properly designed, the phase shift is linear with frequency, and thus translates into a pure time delay. This has several ramifications:

1) The phase shift doesn't "color" the sound. A time delayed signal sounds just like the original, just later.

2) An FIR filter is not a minimum phase system. Thus, it will not properly correct for minimum phase frequency response variations in a system component (a speaker or microphone, for example). You can correct the amplitude response of a speaker with a linear phase FIR filter, but the phase response will still be messed up. Conversely, a good old analog EQ can and will fix both the phase and amplitude response at the same time, as has been pointed out below.

3) FIR filters generate serious amounts of delay. A 1000 tap filter sampling at 44Khz will insert about 23 mSec of delay, which is equivalent to moving your speakers almost 20 feet! In many live sound situations, this kind of delay is totally unacceptable. You can reduce the delay by reducing the number of taps, but then the performance of the filter suffers, especially at low frequencies.

FIR filters find their greatest use in the digital-analog conversion process in digital audio playback. There the delay is irrelevant, and the linear phase response of the low-pass FIR filter is a decided advantage in sonic quality. For live sound use, though, you have to think of them first as primarily delay lines, that happen to have non-flat or selectable frequency response. Sort of a special purpose tool, requiring sophisticated and intelligent application.

Regards,
Ron Riedel

Reply posted by Mikael Holm on December 11, 2001
- "A 1000 tap filter sampling at 44Khz will insert about 23 mSec of delay, which is equivalent to moving your speakers almost 20 feet!"

Many vented boxes exhibit longer group delays at the tuning frequency. Actually that's what i'm correcting with own lil' box. It just needs so much time to work it out :)

Miffe