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BadMouth:

--- Quote from: CoryBee on December 13, 2012, 07:53:54 pm --- :dunno

--- End quote ---

Human perception of sound changes at different volume levels.
The response curve flattens out as music gets louder.\
But at low volumes you hear the mids (range of human voice) a lot more than the highs or lows.
What Loudness Equalization does is boost the high and low frequencies at low volumes and turn them down as the volume is increased.
It will sound clearer to you at lower volumes and your brain is tricked into thinking it's louder than it really is.
Conversely, if highs and lows are cut when it is loud, you will perceive it as quieter than it actually is.

In a way, I guess it is like compressing it on the fly.
I might be happy with that if I had a speaker that had limited range anyway, but I wanted to make sound quality one of the focal points of my cab since it would be playing newer games that have some great music.
BadMouth:

--- Quote from: rCadeGaming on December 13, 2012, 08:09:24 pm ---I took a look at your build thread.  The speaker box you integrated into the cab is pretty impressive.  I'm planning the exact same idea, but maybe with a sealed box instead of ported.

This being said, did you "acoustically calculate" the optimum geometry and dimensions inside the box?  If not, you probably won't get a totally flat response.  I'm not saying I could do better; just saying I'd highly recommend a nice analog equalizer to really optomize your sound quality and get the most out of all that work.  It really makes a difference, especially when you have to make compromises to fit things into an arcade cabinet.

--- End quote ---

The upper speaker enclosures are slightly smaller than the original the speakers were removed from.  As far as geometry, there wasn't much choice as it had to be shaped to the available space.  Before building it, I placed a block inside the original enclosures to reduce them to the amount of space available in the cab and the sound quality was still acceptable.

The subwoofer enclosure was designed in WinISD and then I experimented with slightly longer and shorter ports using cardboard ones before installing the plastic one.  According to the software, it was flat with a -3dB around 30Hz.  I no longer have the equipment to test it, but it is very deep and smooth for an 8 inch speaker.  I'm sure that it firing into the wall from 3 inches away has an effect on it, but I'm not analyzing it that far.  I'd just drive myself nuts.  I'd like to finish the cab within a year of starting it (fingers crossed).
rCadeGaming:
Sounds like you certainly know what you're doing, but as you say, compromises must be made in the speaker enclosure due to the space available.

It may be acceptable, but it could certainly be improved with an equalizer.  It would be worth it to optimize things after all that work.  People are giving away pretty nice used analog eq's on eBay these days.  Here's one I picked up recently:

http://tinyurl.com/ao22cjo

I got a nicer one a few months ago, but I can't find it my purchase history because it was too long ago.

Wait, here's one:

http://tinyurl.com/acowzgd

I didn't pay that much for it though.
brad808:
Our hearing response is best at 2.5khz-3.5khz for a normal person. Look at an image of the fletcher munson curve upside down and you will get an easy to look at (in the same sense you would look at a speaker or microphones) frequency response chart. While it does tend to "flatten" out as sound gets louder we are still most sensitive around those same midrange frequencies.


Keep in mind that any "loudness equalization" or "loudness correction", "night mode" on amps - whatever term they use for it are all different forms of compressors. They are all changing the dynamic range of the signal to make the louder and quietter signals closer together. To be honest no matter what form of dsp you enable in windows or an audio manager they will all be processing the signal through their algorithm in there entirety so all frequencies that are present in the signal will be altered in one way or another. Even if for instance you send a signal through an equalizer and boost 80hz by 3db. The entire signal including the other thousands of frequencies will be processed through the dsp plugin and will be changed depending on how good/bad the math is in the plugin. They won't all be boosted by 3db but they will be effected by the quality of the plugin. There are conversions to different sample rates in plugins and all kinds of things that you wouldn't expect to happen but do. That's why there are plugins that are free and plugins that cost litterally thousands of dollars, because they "sound" different. People that are mixing music or post production work choose a plugin that effects the sound in a way that they feel sounds good or has as little "sound" as possible. I'm getting way off track. The point is if you are going to use the windows loudness correction you might as well use a proper compressor that you can control, and will probably not have as negative effect on your signal if you are worried about audio quality.

On another note I've found demul has just awful sound in a ton of the games. A lot of them are by default settings distorting like crazy. Even if you turn down the windows volume control or your speakers amplifier then you are just turning down a clipped signal. So in that situation then yea the only option you have is to go into the service menu and turn it down and hopefully get an unclipped signal.


I haven't looked but you might be able to find a program that automates a volume change on a per program basis? Might be useful for some of the taito type x stuff?
brad808:

--- Quote from: BadMouth on December 13, 2012, 07:33:17 pm ---Model of phone shouldn't matter as long as the same phone is used every time.

--- End quote ---

If you are testing the exact same signal such as a single frequency sine wave every time then this is correct. For signals that are not the same this theory is not correct. So for games, not correct.

I'll give you an extreme example so you understand why. Let's say you have two phones A and B. The program was written for phone A, you are using it on phone B.

Phone A cheaped out on the microphone and used one specifically around the frequencies for human speech because it's a cell phone, say 2khz and by the time you get up to say 6khz the signal response drops by 9db. The creator of the app corrects for this by changing the algorithm to read signals at 6khz 9db "louder". That way they have a good app that reads things correctly and flat.

Now you have phone B which is top of the line with a perfectly flat frequency response from 20hz up to 20khz. Same program, same algorithm. Now you have a program that is changing what it perceives as being loud currently then what your ears are telling you. In other words you could have sounds that your ears perceive to be around the same level and a meter that shows one signal to be much much higher.

You now have the same phone with two signals in different frequency ranges that appear the same "loudness" but read drastically different on the spl meter.

Hope it makes sense.

Sent from my Galaxy Nexus
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