So why not do things as eq makers dictate things should be done?

I am still not trying to break your balls just saying. I have never met a pro sound guy that would agree with your method.

It really is as simple as can be. The 975 sets distances for time alignment and levels for equal output between channels.

The 10 band peq (parametric equilization) now allows us to read and compensate for rings or nulls. A ring is a hot spot where a null is a low. Bass nulls have also been refferd to as a suck out as all the bass is sucked out.

In my room I have the left speaker in the corner which is the junction of two brick walls. The right speaker has the brick wall behind and a panneling wal to the side. To get the levels correct the brick corner speaker is a full db lower than the paneling wall speaker.

REW (room eq wizzard) is a freeware piece of software that when connected to a microphone reads the rings and nulls in your room.

Your rat shack meter is fine for most of what you might need. Specs say it should be good up to 3k with user reports giving it closer to 2k. Most any room issue you may have will be well below 2k.

Sure. EVERY room correction system (Audyssey, Dirac, Trinnov, ARCOS, etc) measures and EQs each channel independently. NONE of them measure or EQ the interaction between channels.

This is very interesting and I've been trying to frame a response for a week or so, but it's very complex.

There's two basic things going on with audio systems: Magnitude Response and Phase Response. Each speaker in your system will have a these responses that vary depending on where you measure. Of course we measure where we listen.

Our systems are also linear time invariant, which means if we play a disc today then tomorrow it will sound the same (unless something breaks). This means that the response of each speaker run individually and added together is equal to the response of both speakers run together.

However when we add responses we have to add the combined Magnitude/Phase response (the Complex Responses).

Now, delay is phase. It has no effect on magnitude for a single source. For example, if we add two equal sine waves that are 180 degrees out of phase we get nothing (perfect cancellation). However the magnitude response of both signals would be the same but the phase responses would be opposite and when you add the complex responses you get zero. (This is hard to explain without equations, but I'm controlling myself)

So if you set the delay and flatten the frequency response on each channel individually then we should be happy.

Except I can think of a couple gotchas.

1. If you don't actually look at the Phase Response of each speaker (Does REW do this??) then you have no way to check if you wired one speak in backwards (180 degrees out of phase). Also most people don't really have a good sense of phase and what it looks like in a graph.

2. While we do our best to make the response flat there might be some interactions between speakers that may cause slight perturbation in the magnitude response. For example, if both your mains happen to have a 0.5 dB bump at the same frequency they would add together and might be noticeable under test. Will it be noticeable with real world material? Who knows?

Testing speakers in pairs would eliminate both those. Actually I would pick one and check the others one at a time against the one.

Yes, REW can generate phase plots for each speaker, as well as split them into separate minimum phase & excess phase graphs, and derive group delay for both.

Great post EEman. I for one would love to see you go deeper into the technical side of parametric eq. I think the ten band peq will be simple to set as long as a person understands what REW is showing for issues.

I do have a question. Does overall phase change with frequency based phase shifts in crossover components?

You also mentioned setting for a flat response. With stereo the common practice is to set a "house curve". Starting at or just below 200hz dial in a gentle slope to +3db at 20hz. Do we not do this any longer?

Thanks.


Yes. Linkwitz and Reily have become quite famous for designing a crossover filter that is not only is flat in frequency but quite well behaved in phase too. What this means for the speaker is that, in the crossover region, they are moving in the same direction at the same time (i.e. in phase).

With digital signal processing becoming affordable for home theaters it became possible to design filters with perfectly linear phase (FIR filter-Finite Impulse Response filter). These types of filters are popular because linear phase implies that the output in the pass band is the same as the input except shifted in time, usually by half the filter length. It's simple to line up multiple filter outputs (say 10) when you reconstruct the signal.

That's the purist view. Infinite impulse response filters can be made to work also and usually require fewer computations.


idk. I understand the signal processing aspects well enough but the human perception of the acoustics is not my area of specialty. I just do what the experts say. The trick is figuring out who the experts are.

Wow! I REALLY enjoyed the last 3-4 replies.

They struck serveral cords inspiring comments & questions, like 2nd order Butterworth versus 4 LR. Finishing up my coffee break. Later tonight I may have time to process and post. However w/a wedding Saturday and 10 hour days @ AT&T in Georgia I doubt it.

The 975 Destroyer, Tony