Measure your room acoustics and digitally compensate for it ...
Aug 29, 2007 at 5:06 PM Post #16 of 35
Quote:

Originally Posted by bigshot /img/forum/go_quote.gif
It isn't difficult at all. You just start from where the automatic equalization or your test tones tell you is flat. That's your jumping off place, not your end destination- machines are never totally perfect at this sort of thing. Then you listen carefully to a bunch of different music under a bunch of different conditions. Make minor adjustments and see how they work. It probably isn't going to take a big tweak, but with EQ, 2 or 3dB here or there can make a BIG difference. Tweak for a few weeks and you'll find a really nice place.

See ya
Steve



Well said. You can always compare it to the auto settings and go back to that preset. Also, Steve, X2 to your posts on the cable thread, I couldn't bring myself to post on that one.
 
Aug 29, 2007 at 7:59 PM Post #17 of 35
Quote:

Originally Posted by bigshot /img/forum/go_quote.gif
That's sales pitch. You can get to the exact same place with a 31 band equalizer. It's a lot cheaper than the fancy systems but it requires thought and careful listening to make it work properly. It isn't just pressing a button and bingo.

See ya
Steve



Are you saying that the DSP implementation does not create the same phase shift issues as does a multi-band analog EQ? Even before digital solutions were available, analog EQ or and tone controls seemed to fall out of favor due to concerns over phase anomalies that they introduced.
 
Aug 29, 2007 at 9:23 PM Post #18 of 35
Quote:

Originally Posted by Sisyphos /img/forum/go_quote.gif
I also know some people who use Computer Aided Room Acoustics (CARA).
It's fine but you normally have to buy at least some (ugly) absorbers to make real use of it.



That is a fact...
 
Aug 30, 2007 at 1:39 AM Post #19 of 35
Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif
Are you saying that the DSP implementation does not create the same phase shift issues as does a multi-band analog EQ? Even before digital solutions were available, analog EQ or and tone controls seemed to fall out of favor due to concerns over phase anomalies that they introduced.


Phase isn't an issue. You can A/B a good EQ set flat with bypass and there is no difference at all. The most important thing about an equalizer is the ease and simplicity of adjustment. Whatever you use, it should be quick and simple to make minor adjustments. That way you can find the right setting.

Achieving flat response is very difficult. It's a time consuming task. Even a slight imbalance can cause problems up the scale because of masking issues. But the clarity and balance of a flat system is wonderful to hear.

See ya
Steve
 
Aug 30, 2007 at 9:04 AM Post #20 of 35
I also use TacT. I would never go back to an un-eq'ed system. EQ is particularly essential if you do your listening in a normal room where it is not practicable to put in much acoustic treatment.

Anti RC sentiment is mainly luddism. I expect that as computer based listening continues to gain market share more units like the TacT DAC-EQ-Pre are going to appear, maybe something like a Transporter with room correction.

Corrected systems can sometimes seem bass shy at first listen, but if you give it a few hours of listening time you might find that you are just missing the boom you used to have.
 
Aug 31, 2007 at 2:53 PM Post #21 of 35
HT_and_Music_Room_C.jpg


That's a great home theater! And you're right, the absorbers are not ugly at all. If I had a special room to do something like that ...

Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif
The big question is "What target response are you seeking?" The TacT system computes nine different corrections with various bass boost and hi roll-off so you can get an idea of what works right for you. Indeed, a flat system does sound "flat" (meaning, sort of lifeless/cold), because of the non-linear response of the ear and the fact that recordings are mastered to sound "right" to someone, somewhere, who doesn't have a flat room either.


Yes, one really needs to play around with the target response. I really found everything set to 'zero' resulted in a somewhat lifeless sound.
So it's great that you can set the target response as you like with ARC. And once you have done one measurement you can calculate as many FIRs as you like.

Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif
No doubt, ARC is cheaper. OTOH, you can get a used TacT RCS 2.0s for $1200 or so that includes a DAC and preamp, so it's not that large of an investment.


That's true, but if you want to try another DAC / preamp you are still unable to sell the TacT DAC / amp because you need it for the processing purpose.

Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif
Interestingly, TacT suggests that the user investigate less toe-in of the speakers at some point to optimize the size of the "sweet spot" post-correction. The claim is that there is a broader range of more consisent response a bit off-axis than directly on-axis for most tweeters. While it might not measure as flat per se, it's a wider window, and it's going to be corrected, anyway.


I didn't know that but I don't expect that this works with my monitoring-speakers because they have a quite small sweet spot and therefore really need the usual 30 degree toe-in.
But maybe I should just try it ...
 
Aug 31, 2007 at 3:02 PM Post #22 of 35
Quote:

Originally Posted by bigshot /img/forum/go_quote.gif
It isn't difficult at all. You just start from where the automatic equalization or your test tones tell you is flat. That's your jumping off place, not your end destination- machines are never totally perfect at this sort of thing. Then you listen carefully to a bunch of different music under a bunch of different conditions. Make minor adjustments and see how they work. It probably isn't going to take a big tweak, but with EQ, 2 or 3dB here or there can make a BIG difference. Tweak for a few weeks and you'll find a really nice place.

See ya
Steve



Yes, but I am still skeptical that it's possible to make it sound perfect in the whole room. The frequency responses in different places in a room are just too different, therefore it's always a compromise. But of course it can be a really thought-out compromise that allows you to experience good sound in the most important listening areas. Most people are not sitting in the corner while listening to music ...
 
Aug 31, 2007 at 3:10 PM Post #23 of 35
Quote:

Originally Posted by widmerpool /img/forum/go_quote.gif
I expect that as computer based listening continues to gain market share more units like the TacT DAC-EQ-Pre are going to appear, maybe something like a Transporter with room correction.


Most people have not yet realized what DSPs and computer audio are able to deliver. But I still think that when these things gain more popularity devices like the TacT will become more and more obsolete because the computer will take over the TacT's processing job.
 
Aug 31, 2007 at 3:36 PM Post #24 of 35
Es tut mir leid, ich spreche kaum Deutsch........so I can't even tell from the ARC site what files I would definitely need to download.

Can one get through an initial correction without knowing German very well?
 
Aug 31, 2007 at 3:56 PM Post #25 of 35
I'm using the Inguz DSP plugin for the Slimserver (Slimdevices players).
http://inguzaudio.com/

So if you have a Squeezebox or a Transporter it can be upgraded to a room correction device as well.

Works great, has enough pre-set filters for you to choose which one sounds best to you. Only thing that isn't so easy at first is the proper recording.
The mic positioning, mic preamp, all sorts of cables...but it's worth it. There is a guide on his site as well.
 
Aug 31, 2007 at 4:02 PM Post #26 of 35
Quote:

Originally Posted by sejarzo /img/forum/go_quote.gif
Es tut mir leid, ich spreche kaum Deutsch........so I can't even tell from the ARC site what files I would definitely need to download.

Can one get through an initial correction without knowing German very well?



Das klingt doch schon gut ... :wink:

I don't have the time right now but I will send you a short step by step guide with screenshots and a translation if you like - then it should be very easy ...
 
Aug 31, 2007 at 5:13 PM Post #27 of 35
I think a big problem with the flat sound after auto EQing is the response of the microphone sampling the frequencies. I've compared auto EQ systems to ones adjusted carefully by ear with test tones, and ears seem to hear differently than micophones.

See ya
Steve
 
Aug 31, 2007 at 9:45 PM Post #28 of 35
I have used the dBx 10 octave equalizer unit for many years now, which does much the same automatic equalizing. It uses pink noise and can be set to do each speaker separately. DBx recommend doing several test runs with a mic in different locations, so that you are not just optimizing the sound for one narrow location in the listening room. The dBx system allows averaging of several runs. I then do my own adjustments, since you do not know if you can fully trust the unit or microphone. Generally you want to roll off the treble, because there is a natural treble loss in a hall and because a lot of rock pop has hyped treble. dBx discusses this in their manual and the unit has a 4dB/octave filter which can be switched in.

The effect of good equalizing is well worth the cost. The speakers and to some extent the room sort of fall-away and you just hear the musicians. Music which seemed confusing and or wrong
now starts to sound right.

Actually a lot more people use equalizers/tone controls than you would think. I was surprised that in the Stax thread, where you are dealing with some of the most expensive phones and headphone amps, a number of people admitted using tone controls/EQ even on headphones.
 
Sep 1, 2007 at 9:43 PM Post #29 of 35
The ARC website as well as the program's user interface are only available in German so far.
So I am gonna give you the most important facts and try to explain how you will get it work if you want to.

Let's start with the facts:

You will need:

- a Windows PC (I run it on XP and I know that it also works on 2000 but I am not sure about Vista)

- a soundcard with a microphone-in that's of reasonable quality.
If the soundcard's mic is not adequate you will also need a separate mic-preamp. My SC is a Tascam US-144 and its mic-in is alright.

- a measurement microphone
(mine is a Behringer ECM8000)

- a mic-stand - of course

- 99 EUR to spend on an ARC license


Alright, now I will briefly explain how you will get it work:

- Download the software at 'www.high-end-manufaktur.de/arc.html'
To do this click on 'Downloads' in the right column, then select the latest version: 'ARC Installation V.1.1 (mit Kanaltest)' [you will not need the patch right now].

- Now unzip the folder and install ARC.

- Run 'ARC console' and you'll see the ARC window.
Before you can do the first measurements you gotta acquire the license.
To do so select 'Lizenz anfordern' (request license).
Now a new window will open automatically and show a file named 'usdef.zip'.
You will have to e-mail this file to the developer of ARC: 'info@high-end-manufaktur.de'. His name is Ralf Stegmueller and he works at the computing center at the Karlsruhe University.
He is also being very helpful if you have any questions about ARC or digital room correction in general.

- Ok, when you paid for the software and you received the license you'll have to copy the liecense file to the ARC folder (c:\ARC).

- Now you're ready to use ARC.
Place the measurent mic at your listening position, hook it up to the soundcard. Set the playback volume to a rather high level of 85-90dB (a SPL meter is helpful but not necessary - you can just try it). The Tascam's mic input level should be set somewhere between 3 and 4 o'clock (that's about 80 - 90 percent) - I didn't try it with other soundcards or mic preamps.

- You can choose between measurements at 44.1, 48 and 96KHz.
With the Tascam I only tried the 44.1KHz - 96 didn't work but I don't think that it will have any advantages.

- Ok, once you selected the sampling rate it's your choice again: soft, normal or strong correction. They are called 'ARC 44 KHz light / normal / strong'. I think that's somehow self-explanatory. You should start with 'normal'.

- Now, let's go: hit the button 'ARC 44 KHz normal' and be quiet ...
A blue window pops up and soon after you should here a 120 second sine sweep from your left speaker. After this some calculations proceed and then you will here the same sine sweep from your right speaker. Then some final calculations and a new window opens: 'ARC Korrekturarchiv' (ARC correction archive) that's containing your correction files in stereo and mono. The hole procedure should take no more than 8 minutes (depending on how fast your computer is).
That's it.

- Open your convolver (e.g. Foobar's) and load the correction file.
(Be sure to slightly lower the gain [about -4 to -6dB] otherwise you will experience digital clipping - especially with most of the newer recordings.)

Now listen to some of your favorite tunes ...


Hope that short explanation helps you if you're interested in using ARC.

- Marco
 
Sep 1, 2007 at 10:00 PM Post #30 of 35
How do they calibrate for the response of the mike so you aren't correcting for that along with your speakers? Different mikes sound quite different.

See ya
Steve
 

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