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What does room correction software correct?

Making quantitative measurements is useful to "balance" the room; however, the devices involved are attempting to achieve a true measurably "flat" response across the frequency spectrum. The human ear does not respond like a calibrated microphone--refer to the Fletcher-Munson Curve https://en.wikipedia.org/wiki/Fletcher–Munson_curves for information regarding the typical response of human hearing. As you will see from that data, the human ear does not have a flat response across the frequency spectrum, and varies with volume level--that is the purpose of loudness controls for lower level listening. Also, all ears are not created equal--shape/depth of ear canals, continuous exposure to harmfully loud noise levels (like occupational factors), and simple aging has a dramatic effect on hearing/hearing loss. And then there is just the matter of personal preference (which may or may not be related to the previously mentioned factors).

So although a measureable flat response is a good baseline to start with, it may not (probably will not) sound "right" to you. That is the purpose of using filters/EQs/tone conrols/loudness controls to adjust the response curve to your personal requirements/preferences. The technical aspects of achieving a flat response will correct for room factors (reflective points, null points, speaker positioning issues, and resonance), but will not correct for your own ears' response or personal preferences--that is why it is best used as a baseline upon which to make further adjustments/compensation.
 
savatage1973,

Thanks for your great explanation.

It just seems counter productive to start with a flat response that corrects for room factors, but then use controls that adjusts sound to the sound that you like best which in turn affects the room factors that you are trying to correct. Do you know what I'm saying?
 
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It all depends on the acoustic of the room and the capability of the system as to how a room is voiced. If you have very directional speakers or sit close to you r speakers where the room has very little effect then a flatter curve might be enjoyable. Where the room contributes a lot to the sound then roll off is very necessary. Remember you can only voice the sound of the speaker. You can voice the sound of the room only by changing the sound balance of the speaker. So if your room has issues changing the speaker will never correct the room satisfactorily . The room has to be the best it can be first. In my situation I voice my room below 1000 hz or so from the listening position and then voice the speaker at 4 ft distance above 1000 hz and let the room do its thing. Then I go back to the listening position for final minor touch up between 250 and 5,000 hz leaving the highs alone. Remember the ear is marrying two sounds. The direct sound of the speaker which arrives first and then the reflected sound of the room which arrives later. If the room is to soft elevating the sound to make things flat will make the initial arrival sound irritating. Like wise if a room is very hard the room will make the overall sound hard and very ambient in sound quality covering up definition from the direct first arrival of the speaker. Its all very complex and something that takes a lot of reading an experience to get a satisfactory result. Go to live un-amplified concerts to get a feeling of what real sound is. Then pick recordings of faithfully recorded sound of similar performances as a reference. Don't use studio recordings where everything is make believe, for a reference.
 
Somebody please explain, what's the point of taking measurements to make a flat frequency response, but then using filters that is not flat or adjusting frequency to what sounds best to your ears?

Imagine a 10K race, where every runner has a different starting position.

You're basically leveling the playing field.

(Uh ... sorry ... all out of sports analogies for now) <G>

Yes, you will probably want to tweak your basic tone controls a bit here and there for various media sources, either compensating for bad mastering or minor differences in what your source components are delivering. No different than having to crank up the volume a bit for the turntable as most people do. In any case, with good EQ applied to the system and room, you know exactly where you're starting, and it's easy to get back to that point when you're done with that particular source or material.

PS ... don't expect a newly EQ'd room to sound perfect right off the bat. GooOOoogle "psychoacoustics" and you'll find that it takes a while for the ear to adjust to what you're hearing. You'd be amazed how bad what you've been perfectly satisfied with will sound later on, and how much music you've been missing once your retrain yer brain.
 
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savatage1973,

Thanks for your great explanation.

It just seems counter productive to start with a flat response that corrects for room factors, but then use controls that adjusts sound to the sound that you like best which in turn affects the room factors that you are trying to correct. Do you know what I'm saying?

I know what you're saying and agree with you. Since the late 70's I've used measurements and an equalizer (analog then DSP now) to flatten in room response. I had tone controls that I rarely used until I traded an ARC SP-3 in for an SP-9 (no tone controls) in 1988. I've never felt the need or had the desire to act as an editor and second guess the engineer by constantly adjusting an equalizer or tone controls. A lot of my attitude in this area is heavily influenced by the music I primarily listen to. Classical and Jazz are for the most part well recorded.

As for adjusting my stereo to compensate for whatever hearing deficiencies I may have? A long time ago I asked my self, if I do that what would I do about the rest of what I hear. My system is set to be relatively flat at ~87dBA with a slight high end roll off at my listening position. I have no tone controls and a parametric DSP is not the sort of thing one adjusts for every recording. The one thing I would love to have is a true loudness compensation control. Although there are newer more accurate plots than the old Fletcher Munson one, I'll take whatever I can get. BTW: a variable loudness control is one of the main reasons I like current Yamaha gear.

The truth is, when my system and most especially room and speakers and their positioning became better my desire to be an editor with everything that I played pretty much went away.
 
JoeESP9,

I guess I'm in the group that prefers not adjusting tone controls and leaving it flat. My current preamp doesn't have tone controls. I too want to leave the engineers alone with their recordings. My active crossover is adjusted flat. Also, I don't even use my parametric EQ. I used to have a vintage Yamaha R-9 receiver. It was an awesome receiver with variable loudness. It was quite expensive way back in 1987.
 
Also, I don't even use my parametric EQ.
While no amount of EQ can correct room nulls, I find parametric EQ useful to attenuate peaks in the bottom couple of octaves to provide smoother bass response.

I correct two separate peaks on either side of 100 hz with the parametric on my HT processor.
 
The issue - described as simply as possible - is that a small room built of reflecting material is a hostile environment for the development of music and harmonics both in terms of size and time. This causes irregular frequency response (low bass cannot be fully developed and certain low frequencies trigger resonances in the room) as well as poor ambience due to early reflections being summed up with the original sound.

Can't really solve the problem, just reduce it's effect, and depend "on two ears and a brain" - as Floyd Toole likes to describe - to balance out the irregularities.

Room correction curves and dsp tools can perhaps reduce the frequency response issue in one sweet spot that has been measured and optimized. Cannot really do much in the time axis.

There's a school of thought that treating the room properly and carefully placing the loudspeakers (many of them) goes much further in helping build an environment that has fewer defects for your brain to fix.
 
FWIW, a blurb from Audyssey...

... there is no way to control the sound after it leaves the speaker. However, we can measure the effects of the reflections that arrive after the direct sound by looking at the time domain response. It has a certain pattern to it that will depend on the time of arrival of those reflections. Based on that pattern and the similarity of patterns across multiple measured locations we can identify the problems caused by reflections. Then, a filter is created to invert those problems as best as possible.

The key is to not think of the filtering in the time domain. It's not like a graphic equalizer that can only raise or lower the amplitude at certain frequencies. Our filters are in the form of impulse responses that operate on the audio signal through an operation called convolution. As such, they are affecting the signal in each channel in the time domain--hence the name. An additional benefit of such filters (also called FIR) is that they operate in the time-frequency domain as well. That means they can be used effectively to lower the ring down time of room modes in the low frequency range.
 
JoeESP9,

I guess I'm in the group that prefers not adjusting tone controls and leaving it flat. My current preamp doesn't have tone controls. I too want to leave the engineers alone with their recordings. My active crossover is adjusted flat. Also, I don't even use my parametric EQ. I used to have a vintage Yamaha R-9 receiver. It was an awesome receiver with variable loudness. It was quite expensive way back in 1987.

While no amount of EQ can correct room nulls, I find parametric EQ useful to attenuate peaks in the bottom couple of octaves to provide smoother bass response.

I correct two separate peaks on either side of 100 hz with the parametric on my HT processor.

MY DSP is used only in the subwoofer feed which is from 85Hz down. I resorted to it when room treatments and judicious subwoofer placement were unable to correct some bass anomalies. I've run the auto calibration software on my surround pre/pro. The truth is I've paid little attention to the resulting changes. As I posted earlier I have a "good" two channel system to which I added surround. Compromising my two channel sound is not acceptable.
 
Room correction curves and dsp tools can perhaps reduce the frequency response issue in one sweet spot that has been measured and optimized. Cannot really do much in the time axis.

It's possible to deal with timing ... just a whole lot more work, and some level of compromise comes into the equation. REW deals with that by providing a rather impressive set of tools to create "waterfall plots" ...

http://www.roomeqwizard.com/help/help_en-GB/html/graph_waterfall.html

waterfall.png


These track room resonances, decay, and reflections over time accurately and give you a good idea of what's happening and where your hardscaping needs work. I believe jRiver also has an addon that can incorporate these results in the DSP settings - don't quote me on that.

Once again, I'll leave this sort of stuff to the fanatics. Not really an issue here as the only place I'm concerned with is the great comfy chair. Adequate is more than good enough for the rest of the house as that's only background music. Still might be fun to play with come winter when the walls start closing in ...

Hmmmm ... Wonder what effect walls closing in would have on the plots? ;-}
 
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It just seems counter productive to start with a flat response that corrects for room factors, but then use controls that adjusts sound to the sound that you like best which in turn affects the room factors that you are trying to correct. Do you know what I'm saying?

It is not necessarily "counter productive", it just adds a little more work to the equation. If you have a room with windows on one side (highly reflective), or an open wall on one side (dissipative), or exposed brick or stone (rather than drywall/plaster) on another side. or if one speaker is less than a foot away from a side wall and the other is six feet from a side wall, the room dynamics play an important role in how each channel (speaker) responds/sounds. By setting a baseline (compensated for each channel as flat), then you have a "level playing field" for each channel to make further adjustments/compensations for what sounds "right" to you. Most homes are not designed with dedicated listening rooms or anechoic chambers that are optimized for audio performance (I know my 130+ year old house is not), so if I can "balance" the room by "flattening" it first, then I can move on to what works for me.

I get what you are saying--just set it where it sounds good to you, and leave it alone (and that is a lot easier), but your channels may never be truly "balanced"--one channel may always sound "brighter" or the other "bass heavy"--you never know until you "level/flatten" them as a baseline to make them as "equal" as you can reasonably achieve. Most people tend to set the bands on an EQ identically for each channel, where in reality, they may need to be totally different curves for each side to achieve true "balance".

Hope this makes sense, sorry for the rambling...

EDIT--just to throw some more "reading material" into the equation to further complicate matters, give this a read http://www.cardas.com/room_setup_main.php
 
Most homes are not designed with dedicated listening rooms or anechoic chambers that are optimized for audio performance (I know my 130+ year old house is not), so if I can "balance" the room by "flattening" it first, then I can move on to what works for me.

An anechoic chamber/room is a truly awful place to listen to music. Likewise with an overly damped room, unless listening in the near field.

You need to strike a reasonable balance between absorbing, reflecting and diffusing.
 
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