• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Room correction, speaker correction anything above Schroeder a mistake?

From what I'm reading people are saying it may be pointless to EQ speakers above a room's schroeder frequency? I currently cut of auto EQ at 500Hz in my room (EQ'ing only below 500Hz). Should I instead try to estimate the schroeder frequency for my room and place the cut off at that frequency instead? And why?

Thanks.
Pretty sure there is an existing thread with almost exactly this title. Suggest browsing for it.
 
Pretty sure there is an existing thread with almost exactly this title. Suggest browsing for it.
After looking harder I found this.


I'll ready though it.
 
One key issue is how the measurements are made. The resolution of our hearing progressively gets lower and lower as frequencies go up so the frequency response needs to be measured that way. Without it, you will be fixing troughs and peaks that are not audible.

The other point is that the speaker dominates the response in higher frequencies. So poor response there is best fixed using a better speaker.
Since our hearing gets poorer as the frequency increases would you recommend changing a REW frequency graph reading to variable so that the peaks and troughs become more of a psychoacoustic estimation in the highest frequencies?
 
I think the more apt descriptor is that it's "tricky" to EQ above Schroeder -- not counting the more common broad type HF shelving filters nearly all studio professional monitors have built-in.
 
Pretty sure there is an existing thread with almost exactly this title. Suggest browsing for it.
Thank you @Jimbob54 for pointing out the duplicate thread. Threads consolidated. ;)
 
EQing above Schroder is useful to get L/R speakers to match better, in the presence of room interactions. This directly impacts imaging performance; the better the speakers are matched to one another, the more precise the imaging. This is quite obvious, IMO; if L/R is 100% identical and a tone is played, you get a perfectly centered image. If one channel is louder than the other evenly, the image shifts towards the louder speaker. However, if one channel is unevenly (across the freq range) louder/quieter than the other, then the image smears; some portions of the sound go towards one speaker, and others go to the opposing speaker. Room interactions/reflections can affect this behavior for the different speakers, and room correction could be used to reduce this effect.

As to meeting a particular target curve or correcting some phase response, I'm not quite as sold on that part, other than for some basic tone control/correction (i.e. tame a bright speaker).

Below is some analysis I had done when I was evaluating Audyssey XT32 and Dirac Live in my system. Both DRCs improve L/R matching vs. no correction for SPL. Each line is the average of 6 measurement points around the MLP.


1649970615366.png


1649970626350.png


1649970639389.png




Additionally I looked at L/R phase error, which is also something that is audible (i.e. flip the phase of one of your speakers, you'll hear a difference). DRC improves upon it, although I found Dirac doing something weird here and increasing the error in the mid range. Subjectively I did not like the way Dirac sounded above the Schroder frequency; the imaging got more precise but the soundstage sounded like it reduced. Audyssey, on the other hand, sounded better on than off above the Schroder frequency. The average error numbers are calculated on linear scale, but when viewed in log scale, it looks like Dirac is worse than No Correction.

1649970819283.png
 
Last edited:
I am not sure we are talking about the same thing. The topic to search for is Critical Bands. This is the bandwidth of auditory filters. See https://en.wikipedia.org/wiki/Critical_band

400px-Band-pass_filter.svg.png

What we are discussing here is the difference between F2 and F1. As the center frequency goes up (Fc), that differential gets wider. That relationship is indeed linear with frequency. See this for example from Zwicker & Fastl, "Psychoacoustics, Facts and Models" book:


View attachment 9846

Above 500 Hz the relationship is almost linear.

In lay terms then what this says is that we cannot discriminate sharp drops and peaks at high frequencies as we do on low frequencies. So we better filter our measurements to reflect the same.

For this reason, when looking at low frequency response of the room, we want to use no filtering or 1/12 octave or better. At the extreme of our hearing, we probably want to be at 1/3 to 1/6 octave to roughly match the discrimination of our hearing.
The bolded part is triggering a thought...

Why do we care so much how a speaker measures over 500 Hz if our hearing cannot perceive the "drops and peaks" above that frequency? I might be missing something, but is the linearity of the speaker above the Shroder frequency going to be 'lost' on my hearing? If so, shouldn't we worry more about the linearity of the speaker below said frequency (and I don't mean worry with DSP... I mean in general). I am not suggesting the curve is meaningless... just that small variations that we would have trouble discerning might also be a similarly small variation of different speaker measurements over the same frequency range.

And I also can't hear over 14,500 Hz... so add that into the mix.

:confused:


Thanks.
 
The bolded part is triggering a thought...

Why do we care so much how a speaker measures over 500 Hz if our hearing cannot perceive the "drops and peaks" above that frequency? I might be missing something, but is the linearity of the speaker above the Shroder frequency going to be 'lost' on my hearing? If so, shouldn't we worry more about the linearity of the speaker below said frequency (and I don't mean worry with DSP... I mean in general). I am not suggesting the curve is meaningless... just that small variations that we would have trouble discerning might also be a similarly small variation of different speaker measurements over the same frequency range.

And I also can't hear over 14,500 Hz... so add that into the mix.

:confused:


Thanks.

Well, the 'drops and peaks' aren't audible with broadband signals,but if I sweep a sine slowly over the peak, you'll notice! There really are many more constraints than show up in most discussion. The issue is signal dependent to an extreme.

I'm giving that talk on room design this Wednesday for the AES local section https://pnw.aessections.org/ I won't talk about the correction part in great detail,but you can read about that in much older talks by myself and Serge Smirnov (which you can find in "meeting recaps" in the old site for now (www.aes.org/sections/pnw) although that will eventually vanish.
 
Thanks, but it is a little lost on me. I did read it, but I read "smoothly equalize the direct arrivals at high frequencies". I am assuming that high frequencies are above 500 Hz.
 
Thanks, but it is a little lost on me. I did read it, but I read "smoothly equalize the direct arrivals at high frequencies". I am assuming that high frequencies are above 500 Hz.

More like 2kHz. It's a hearing question more than an acoustic question, and there are some very notable issues about the "schroeder frequency" idea, as well.
 
More like 2kHz. It's a hearing question more than an acoustic question, and there are some very notable issues about the "schroeder frequency" idea, as well.
So then we should be correcting above 500... and above 2k? In the manner noted in the recaps?

Thx.
 
"The issue is signal dependent to an extreme." tells me...it's complicated.
 
  • Like
Reactions: j_j
So then we should be correcting above 500... and above 2k? In the manner noted in the recaps?

Room correction and speaker correction are not the same thing. If you can get a quasi-anechoic measurement of the speaker, measure the listening window, etc. then you can go for it all you like all the way up to 20kHz. But once the speaker is in a listening room and the microphone at the listening position, only the bass can be corrected and the top end adjusted with a gentle tilt.
 
Room correction and speaker correction are not the same thing. If you can get a quasi-anechoic measurement of the speaker, measure the listening window, etc. then you can go for it all you like all the way up to 20kHz. But once the speaker is in a listening room and the microphone at the listening position, only the bass can be corrected and the top end adjusted with a gentle tilt.
Again... sorry if I am beating a dead horse, but this goes back to my original question. Why is there no correction over 500 Hz? If we can't perceive the nuanced differences that the mic can hear, why do we care about the linearity of a measurement (within reason)? If we are close and we can't notice the slight variation, what does it matter how linear our result is if we can't really distinguish minor variances? Now... if the answer is, it is signal-dependent and you can adjust minorly above the Schroeder frequency, then ok. It is just very veiled and difficult to understand.
 
Room correction and speaker correction are not the same thing. If you can get a quasi-anechoic measurement of the speaker, measure the listening window, etc. then you can go for it all you like all the way up to 20kHz. But once the speaker is in a listening room and the microphone at the listening position, only the bass can be corrected and the top end adjusted with a gentle tilt.

I'm afraid we must disagree, unless your "gentle tilt" is more than 1 or 2 dB per ERB.
 
Room correction and speaker correction are not the same thing. If you can get a quasi-anechoic measurement of the speaker, measure the listening window, etc. then you can go for it all you like all the way up to 20kHz. But once the speaker is in a listening room and the microphone at the listening position, only the bass can be corrected and the top end adjusted with a gentle tilt.
Isn't the tilt just a product of the easier absorption of higher frequencies in normal rooms, and the narrowing of dispersion in most speaker designs as frequency goes up?
Because when I measure my speaker in the middle of the room - quasi anechoic - and EQ/filter it flat. I get a gentle tilt, as I put the speaker back in place and measure in my listening position.
I just read JJ's presentation. Makes mostly good sense to me, even though I only used IIR filters in my own system.
 
The room curve with the tilt is simply what listeners tend to prefer at the main listening position.

How you get to it, is up to you.

If the speaker already measures 'flat' and otherwise well in a Klippel/chamber, it's what you get at the MLP if the acoustics of your room at the MLP are optimal.

That's my simple understanding.
 
Back
Top Bottom