Can anyone help with my DIY room correction process ?... (1 Viewer)

[Jones_Rush]

Hi,
I am running my audio system from a HTPC. I have an unlimited band Parametric Eq software (which uses fir filters and not IIR filters, so you can use it also with the main speakers and not only with the subwoofer). I would like to use this software in order to apply room correction for my listening spot, but only below 200hz (maybe even lower).

The question is this: what is the best process to do it ?. Should I calibrate both speakers at the same time (using the same signal) ?, or, should I calibrate each speaker individually ?.

I know that when it comes to professional RCS systems, they calibrate each speaker individually (since high frequencies are involved, meaning, both speakers won't play it at the same time). OTOH, I know that regarding dual subwoofers, the notion is that you need to calibrate them together at the same time (using the same signal) because in real life they will be playing the same signal at the same time (since low frequencies are non directional). So, what do you think I should do in my situation ?.

 

[Jones_Rush]

Anyone ?.

 

[Wayne A. Pflughaupt]

I EQ’d mine separately first, then turned them both on for the final tweaking.

Regards,
Wayne A. Pflughaupt

 

[Jones_Rush]

I EQ’d mine separately first, then turned them both on for the final tweaking.

Wayne, let me get this straight, you first eq'ed each speaker individually to be flat at the listening spot. Then, after both were individually flat at the listening spot, you eq'ed them together, so when they both play the same frequency at the same time, it will sound flat at the listening spot ?. But this way, they will no longer be flat individually at the listening spot, right ?, if so, what will happen if you play a stereo signal which plays a 80hz signal JUST from the left speaker ?, it won't be flat, right ?. Is this kind of a "best compromise" ?, in comparison to ONLY calibrating them individually, or ONLY calibrating them together.

 

[Jones_Rush]

Wayne, I've checked with Tact Audio, and their Room correction products only calibrate each speaker individually, and never together. Am I missing something ?.

 

[Todd Shore]

"...for tweaking."

Small adjustments.

Another reason to at least test them together is to see if cumlative output might excite any resonances.

 

[Todd Shore]

How flat is flat? With a single subwoofer, not very. Half decibel variation would be considered ideal. I wonder how many people are getting there.

With dual subwoofers you can also have cancellation effects so testing a single channel at a time might not give you a good indication of what the summed response is.

 

[Jones_Rush]

With dual subwoofers you can also have cancellation effects so testing a single channel at a time might not give you a good indication of what the summed response is.

I know. But the same is true for equalizing the two main speakers. So this is why I don't understand TacT audio, why aren't their Room Correction devices doing a mutual test for both the speakers at the same time ?, do they know something that we don't ?.

 

[KevinJP]

I have an unlimited band Parametric Eq software (which uses fir filters and not IIR filters

could you expand on this? what software gives you this?
fir? IIR?

tia

 

[Pete Mazz]

do they know something that we don't ?

Speakers are reproducing discrete signals. Subs reproduce mono signals.

Pete

 

[Stephen Dodds]

I do a similar thing to what you are talking about using the dbx Driverack 260. I also used to own a TACT.

I measure and correct each main speaker separately, then do the sub with a BFD.

The most important thing is knowing what you are correcting, which means an accurate measuring system, and getting the right correction values.

There is a very good article on how to do this using ETF 5.0 on the Tag Mclaren website (www.tagmaclaren.com).

You'll find it by going into the VIP Club and checking the TMREQ stuff.

Steve

 

[Jones_Rush]

could you expand on this? what software gives you this? fir? IIR?

IIR vs FIR:
"filters or EQ can be built in two flavors, either Infinite Impulse Response (IIR) or Finite Impulse Response (FIR). Most filter that we interact with on a daily basis are of the IIR type. They trade off a frequency–dependent amplitude boost or cut in return for some amount of frequency–dependent phase shift, also known as group delay. Unfortunately, that’s a problem…although we expect them to operate in the frequency domain, they also muck up the time domain. To understand this group delay thing, think of your basic two way loudspeaker (see figure 8 below). The tweeter and woofer are both mounted on the front baffle. The tweeter’s tiny voice coil is very close to the baffle while the woofers large cone offsets it’s voice coil back quite a bit from the baffle. Now, the voice coil and cone comprise the motor that makes a speaker move air so broadband sound launched from the woofer is offset in time from the tweeter’s output by the physical displacement between the two devices. That’s why many loudspeakers have slanted back-leaning front baffles, which time aligns the drivers.


Figure 8 - An simplified version of group delay via frequency-dependent acoustic delay

This is classic group delay, where the high frequencies from the tweeter arrive at your ear before the woofer’s low frequency content. Think of what this “time smear” does to a broadband sound like a tasty kick drum, a signal with both the HF snap of the beater and the LF boom of the shell’s resonant cavity. Anyway, this same time smear or group delay occurs in all IIR filters to a greater or lesser degree. The only difference is, unlike a speaker, in electronics the high frequencies lag while the low frequencies lead.

As to FIR filter, they exhibit a constant group delay regardless of frequency, so no wonky phase shift problems but, they have a different problem: pre–echo. FIR filters have an annoying tendency to present a small amplitude version of their input at the output, before the input has been applied! I know, what have I been smoking? Nothing, old boy, I simply haven’t mentioned that FIR filters cannot be built in the analog world, only in the bits and bytes of a digital implementation. So, a rip in the time–space continuum that I’ve just mentioned is taken for granted in digital signal processing circles…It’s just one more thing a designer must contend with.

The upshot is that FIR filters, since they lack group delay, don’t “sound” like EQ as we know it. Because of their sonic neutrality, they’re usually used only for specialized correction like in mastering, forensics…or restoration!

I'm using SuperEQ, which is a winamp plug in, which gives me the ability to run an infinte (depends on CPU power) number of Fir filter bands, set at the Q, frequency and amplitude of my choice. Yes, it does suffer a bit from the pre-echo artifact, but it is still miles ahead of what an IIR filter can do in terms of keeping the natural tones of the original recording. Moreover, I'm now learning how to use a MUCH more complicated code (I say "code" because it still is in developmental phases) , which enable me to do a complete room correction with fir filters, and, the pre-echo artifacts are addressed specifically, so they are no longer a problem anymore. Someone who own the Tact RCS correction device, also checked this software recently, and found it about equal to his Tact in performace. He was checking a version of the software which is now much improved. Draw your own conclusions. Anyway, I don't take anyone's word for granted, especially not on the net, and this is why I'm checking it for myself now. The good thing is that it is all for free, but you do need a good mic + a decent sound card + lot's, lot's of free time on your hand, since you need to work with a very bad interface (MS-DOS based), but after you set it once, it suppose to work without any more hassles.


Stephen, the dbx Driverack 260 looks like a real high end piece of gear. How many bands does the parametric eq gives you to work with ?. Btw, why did you stop using your Tact ?.

 

[Terry Montlick]

This is a great question!

If:

1. You are EQing for a single listening position, and
2. that listening position is equidistant from both speakers, and
3. the speakers have identical characteristics, and
4. you are hearing the near-field response of the speakers (i.e., not the room), then

equalized from the same signal, they should have exactly the same filter settings. You can EQ them both together or separately, as long as you adjust the filters the same. It doesn't matter. The same sine wave played through either speaker will have the same phase at the listening position, so there will be no cancellation. And under these conditions, it doesn't matter whether you use IIR or FIR filters.

Of course, if the above 4 conditions are not met, then all bets are off.

I know, this seems like a cop-out, and it is. There's no particular reason for choosing your one channel over the two 80 Hz signal configuration. This situation seems to be a good argument for having a single LF channel (and a high sub crossover frequency). At least you can EQ it rationally!

Regards,
Terry

 

[Terry Montlick]

I simply haven’t mentioned that FIR filters cannot be built in the analog world, only in the bits and bytes of a digital implementation.

More generally, they can't be built in a continuous-time world. Digital is not strictly necessary - only a sampled data representation. There have in the past been FIR filters built out of analog delay lines!

Regards,
Terry

 

[KevinJP]

hm supereq for winamp looks interesting. takes it a good second to take effect huh? will play with this...

what about eq for DVD playback? I have been able to get Zoomplayer to use the directshow eq filters from SonicFoundry Soundforge. are these the good type of eq filters? paragraphic, parametric, plus 10 and 20 band graphic eqs can be used here. anyone know about this?

 

[Terry Montlick]

OK, I'm a little slow tonight, but now I see what Jones_Rush is getting at. An FIR filter can be a linear phase device, so one COULD equalize for any listening position by adding the correct relative delay. You don't have to be equidistant from the speakers, because the right delay correction can be precisely added without messing up phase.

But here's a wrinkle. Assume the signals applied to all the nasty mechanical stuff (speakers cones, speaker enclosures, room, etc.) start off as nice and flat in the purely electronic world. Then the effects of all this non-ideal stuff (room mode resonances, for example) are due to good old-fashioned, causal IIR (albeit electro-mechanical) analog filtering. An ideal equalizer would be an inverse filter that is similarly causal, set up to cancel the poles and zeros of the ugly, real-world processing. But this inverse filter, while appearing to introduce frequency-dependent phase shift, should actually correct it! A typical minimum-phase IIR equalizer is probably a good approximation to this.

And a linear-phase FIR filter would NOT correct the phase problems, but simply preserve them.

That's enough thinking for tonight. I'm going to sleep.

Goodnight,
Terry

 

[Dennis XYZ]

although we expect them to operate in the frequency domain, they also muck up the time domain.

That's not really true. Most "room correction" takes place below 300 Hz where the room and the speakers interact as a minimum phase system. If the room causes an amplitude resonance, it also causes a phase shift. Rather than "mucking up" the phase response, an IIR filter (or an analog filter) "corrects" the phase response to put it back the way it should be. Now, crossovers are another kettle of fish and FIR filters have some advantages (and some disadvantages) there. But, for simple bass EQ duties, IIR filters are the tool of choice.

 

[Stephen Dodds]

Jones,

The Driverack is indeed a nice piece of gear, altho at $679 it isn't really high end in price.

It gives 13 up to 21 bands of parametric EQ a side, although that depends how it is configured, and is also a three way digital crossover with phase and delay adjustments.

For straight EQ the Behringer DEQ2496 might be better, and is much cheaper.

I gave up the TACT because I built my own speakers which need to be actively triamped, and actively EQed before even starting on room EQ.

The TACT doesn't do crossovers and I wanted everything in one box. The TACT is a better room correction system, but the Driverack can fix up the bass enough so that it is close.

Steve

 

[Terry Montlick]

Dennis and I are saying the same thing. IIR equalization filters correct real-world IIR effects, where time and phase necessarily change together.

Regards,
Terry

 

[Jones_Rush]

Terry Montlick and Dennis XYZ,
I am not sure that you are right about IIR filters, maybe in theory you do (or just in basic bass eq duties), but in practice, for full range correction, Fir filters sound way more neutral than IIR filters.

There is another thing: I've contacted more than a few software companies which create software parametric eq's using IIR filters (I had no choice, since I found none who use FIR filters), and asked them what are the advantages of IIR filters over FIR, which made them chose one over the other. What they all basically said is that FIR filters are more appropriate for eq tasks. They ALL agreed that IIR introduce audible phase artifacts, but they also never forgot to mention that the phase artifacs are no worse than the products of the competition. Most of them said that they ARE working on FIR filter usage, and are pretty enthusiastic about its capabilities, when implemented correctly. And just that you won't say that I'm talking in the air, here is one of the email correspondence, which I have currently in hand:

An ideal equalizer would be an inverse filter that is similarly causal, set up to cancel the poles and zeros of the ugly, real-world processing. But this inverse filter, while appearing to introduce frequency-dependent phase shift, should actually correct it! A typical minimum-phase IIR equalizer is probably a good approximation to this.

Terry, you do realize there is a problem here. If the IIR filter would really have done a good job by altering the original phase of a signal in order to compnasate for the altered phase created by the room, then you wouldn't have said that the "ideal" filter would be a *minimum-phase* IIR filter. By saying "minimum phase", you actually acknowledge the fact that the changes in phase introduces by the IIR filter are harmful, and this is why you limit the phase alteration to a "minimal amount", since no phase alteration is not possible with IIR filters, at least not as they are today.

You say that IIR filters are the filters "of choice". This is kind of an injustice , since the advances in computational technology only recently allowed for a real time FIR filter usage (while allowing to use enough bands), so companies didn't really had a "choice" by now. I believe things are going to change from now on.