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Digidesign 192 I/O Outputs too loud?
Old 31st January 2003
Digidesign 192 I/O Outputs too loud?

I am having problems with ProTools hardware inserts.
I haven't calibrated the I/Os of my 192, I am using it as it came. I use the +4 I/Os and when I do a hardware insert, the output of the 192 is too hot for the processor.

Connecting it to a Fatso is the worse case.
Just for reference when I do an insert of a Fatso on the Master Fader I have to turn down the Fatso input (0-10) to 1.5, more than that it starts clipping.

Do you guys have trimmed down the I/Os of the 192s?
Or is the Fatso calibrated diferently from normal?

Old 31st January 2003
7rojo7's Avatar

Calibrate your 192 and match nominal levels with your gear and post again.
Old 31st January 2003
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e-cue's Avatar

Re: Digidesign 192 I/O Outputs too loud?

Originally posted by EduardoApolonia
I am having problems with ProTools hardware inserts.
I haven't calibrated the I/Os of my 192, I am using it as it came.
I agree with 7rojo7 here. Using an uncaled 192 is just wrong. On my TDM system I dedicate the last 888 (a 16 bit one) just for audio triggers and other things that don't audibley make it to the board. I tweek the outputs on these quite often esspecially for 001 clients that bring me sessions with smpte printed onto an audio track for syncing.
Old 1st February 2003
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groundcontrol's Avatar

The problem is that your tracks are recorded close to digital full scale and thus bear no relation to the analog world of 0VU=+4dBu. The 192 i/o comes factory calibrated for 0VU=+4dBu=-18dBFs. That means that when a track meters at -1 on your PT peak level meter the analog signal sent to your outboard insert unit peaks at +17 over 0VU which is it's nominal reference level. Now if this is an uncompressed snare track or some other transient rich percussive sound and the analog unit has reasonably high headroom (likely in the +22db to +26db in pro gear) you're probably not too far off the ideal operating level of this analog box. However if the track is an instrument that has less "attack" and more "sustain", i.e. a sound that has a realtively low "peak-to-average" ratio (aka as "crest factor") like, say, fingered bass or a slow pad and that track is also recorded in PT peaking at -1, then you're feeding the analog unit with an "average" or RMS level that is clearly way over what it wants to get. (You're probably also overdriving the analog stages of the 192 i/o at the same time which is a less than ideal way of making it sound good.) Such a signal should modulate around 0VU on this analog box VU meter if it has one, so it would have to be attenuated by close to 16 dB before being fed to the analog box. The easiest way to do so would be to insert a trim plug-in right before the hardware insert.

If you want to be able to easily interface with the analog world while working with a DAW, you'll have to respect the analog level reference of 0VU. That means that our pad track from earlier would have been recorded reading around -18dB on PT's meter since it has no peaks so to speak. While our SD recorded reading the same 0VU level on the analog box VU meter (which is an RMS or "loudness" sort of meter) would peak a lot higher on PT's meter, probably just shy of 0dBFs (Full digital Scale) in fact.

It's a bit late over here but hopefully it should make sense...

Old 1st February 2003
7rojo7's Avatar

That's the best description of this relationship between the front end, the converters and the monitoring, or for the use as inserts, that I've read. Very non-emotional and practical.
I've always had my 888 set at -12 and it's worked fine, now with the 192 I have to settle for -15, it seems.
I've always been a staunch supporter of hot levels, but I've been recording things for a while and know at what levels to record different types of program material. The problems used to be tape compression, crosstalk and print through, while cross talk remains (on the way in and out on multichannel converters, ce n'est pas grave), most of the problems of tape have dissappeared. I guess this would make some more adventurous in the quest for the holy resolution. Live and learn.
I check my alignment about twice a month, I never had to tweak my 888 after the initial alignment, I've only had my 192 for a few months and haven't been working much lately. But it came in a couple of dB hot on the output. It was the first thing I did when it arrived.
Old 2nd February 2003
Thanks for the (very good) explanation groundcontrol.

So let's think about this situation:

I'll grab your example:
You put a compressor on the master channel to compress the whole mix that is peaking at -1db in PT and has a high average level.

At what value would you advise that I should calibrate a 192?
Will a compressor work well in the (+10db to +17db) range? wouldn't it add distortion to the signal?
What is the best range for a compressor to work?

Would a mastering engineer that uses analog equipment with PT calibrate a 192 to something like -6db?


Sorry for my english.
Old 2nd February 2003
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chap's Avatar

cheap trick

the 192 comes calibrated 2db hotter than most converters. The theory maybe is that louder is better. This is wrong and you need to calibrate your Digi converters.
Old 3rd February 2003
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groundcontrol's Avatar

Eduardo, there is no fast and easy answer as to how to relate a PT signal level to an analog equipment level if you only use the meters in PT because they are "peak" reading meters and each and every type of signal has a different "peak-to-average" ratio.

In digital land, the way signal levels have typically been monitored is to relate them to the absolute loudest signal the system is able to produce and pass on to the converters. This is referred to as 0dBFs (Full Scale). That is when your signal hits 0, that's it, there is no getting higher and hard clipping of the waveform follows.

In the analog world the concept used is different. We talk of a "reference" level of 0VU (Volume Units) which is then referenced to an equivalent signal level and voltage measurement.

For example in the pro audio world that reference is 0VU = +4dBm = 1.228V compared to the consumer standard of 0VU = -10dBV = 0.316V. That means that if we were to feed a sine wave that reads 1.228V on an AC voltmeter to an analog processor with it's input and output level controls set so it's not adding or subtracting any gain (this is called "unity gain"), it's VU meter (which is an averaging meter that relates to how the ear perceives loudness and not the instantaneous absolute peak value of the signal) should read zero. Since a sine wave has NO peaks whatsoever, this same signal would read -18dB (Fs) when fed to a factory calibrated 192 i/o. Hence we say that it is calibrated to 0VU = +4dBm = -18dBFs.

Now, one important notion in the analog world is that of "headroom". This is the amount a signal can exceed 0VU in a particular piece of equipment without incurring "clipping" or distortion and is expressed as +something dB. (It is, in fact, the actual maximum AC voltage an equipment can accept at its input, pass through its circuit and produce at its output without clipping of the waveform.) This vary greatly from unit to unit with the best equipment out there (GML, Cranesong and the likes) having headroom somewhere near +30dB! That means that under normal circumstances these pieces can't really be clipped (unless you add a ludicrous amount of gain like in an equaliser for instance).

How should you calibrate your converters?

This is dictated in part by the headroom of the equipment you are using. If you are using equipment with 12dB of headroom for example, calibrating your system to 0VU = -18dBFs would be less than optimal since these pieces will not produce peaks reading more than -6 in PT before clipping the signal. So they will never clip your converters. On the other hand, why use a calibration level of 0VU = -12dBFs if your equipment is able to pass undistorted peaks of +30dB?

The type of music you record also comes into play in deciding how to calibrate your system since, heavily compressed pop/rock instruments or already "produced" synthsounds and samples most probably will not contain the same peaks (or have an as "high" peak-to-average ratio should we say) than uncompressed jazz, classical or acoustic instruments.

Also, your position is different if you do a lot of tracking of "raw" instruments to be processed with plug-ins inside PT compared to doing mostly mixing with analog hardware inserted on already more produced and "controlled" tracks.

Since I own some high headroom equipment and do both a lot of tracking and mixing of varying musical genres, and since I don't like re-calibrating my system every other song (also since I like overdriving some pieces like some Neve modules at times...), I presently stick with 0VU = -18dBFs and don't fret too much if every other track is not hitting close to zero on PT's meters. (I don't think that it's a good idea to hit PT's summing buss with too much level when mixing anyway but that's another totally different (and much discussed!) can of worms...)

To come back to your example, my guess would be that this pair of converters could probably be calibrated somewhere around 0VU = -14dBFs. When mixing (presuming that you use a pro quality unit with relatively high headroom), I would first put the compressor inserted across your mixbuss into bypass and check the input level (what you're feeding it) on it's VU meters and try to build my mix so that it normally reads around 0VU to +2 with peaks maybe hitting +3 before compression taking place. Again presuming that once you engage the compression the output level would normally kick back a bit and stays around 0VU. For a typical overcompressed pop mix, the peaks are rarely more than 12 to 14 dBs over the average level before final limiting so this should theoretically work.

Of course there are more subtleties to using analog gear that I can get into here. For instance, many engineers find that different pieces have different "sweet spots" as far as hitting them with different levels and will use the way certain equipment starts to behave in an sonically interesting and flattering way before crapping out when pushed a bit. By contrast, certain pieces (cheapy mixers for example) are not happy with "hot" levels and are better operated conservatively. So, in the end, it's still back to your ears and what they are telling you...

Like our beloved Slipperman would say: HO HO HO!!!

Have fun!
Old 3rd February 2003
Thanks Again for taking your time to explain me this well.

Getting to the example again:

Originally posted by groundcontrol

I would first put the compressor inserted across your mixbuss into bypass and check the input level (what you're feeding it) on it's VU meters and try to build my mix so that it normally reads around 0VU to +2 with peaks maybe hitting +3 before compression taking place. Again presuming that once you engage the compression the output level would normally kick back a bit and stays around 0VU. For a typical overcompressed pop mix, the peaks are rarely more than 12 to 14 dBs over the average level before final limiting so this should theoretically work.

So in this case to achieve 0VU to +2 on the compressor before compression and my 192 being calibrated at -14 that would mean that my mix would have peaks of only -14 to -12 right?
Following the example: After beeing compressed it would return at compressor's 0VU. and that would be -14db in PT, right? Isn't that a low final volume for a PT mix?

About the headroom of the compressors:
This all started with the Fatso. I never had any problems with the other compressors.
I read in another post that you have a Fatso. Do you know how much is the headroom of the Fatso? Do you feel it has a small headroom above 0VU?

Thanks again.

If you are getting tired of my questions just say so. I'm a little bit persistent and make a lot of questions until I fully understand and know how people work with it.
Old 4th February 2003
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groundcontrol's Avatar

I don't think you understood the difference between an "average" or "loudness" level and a "peak" level judging by your question. An audio program has both a simultaneous "average" AND "peak" level and those levels are usually gonna be of different values. For example, a drum program that has an "average" level reading around 0VU has typically near instantaneous peaks of 14 dBs or more over its "average" level of 0VU. So if your converters are calibrated so that a level of 0VU reads -14 dBFs on PT's meters, our drum program averaging at 0VU and containing peaks 14 dBs louder than 0VU will read as having peaks at 0dbFs on PT's "peak reading" meters and your converters will be on the verge of clipping.

I would suggest you re-read my previous posts if this is still not clear as I don't know how I could better explain this. (BTW, French is my primary language and not English so that's not helping either...)

Concerning the Fatso, it is capable of a lot of gain so it's normal to have to run it's input very low if you're feeding it a hot signal and you want it to work lightly. (As you would on a mix for example.)

Old 4th February 2003
I think that I understand the difference but maybe the question wasn't very well laid. (my primary language is portuguese so it doesn't help me too)

I'll give you my real example of my problem than you can advise me of what to do.

I have a mix of a Heavy Metal band that is already very compressed to have that big sound, than I apply a L2 to limit more peaks on the mix and get the "average level" louder about 3db, although peaking at the same -0.5db at PT.

Then when I insert the Fatso at the Master Fader I have to turn the Input to about 1.5 so that the pinned led doesn't glow constantly. althought the 0VU led is almost all the time on.

I have also a SPL Tube Vitalizer that when I put on the Master channel it starts clipping if I add something to the signal. I think that all the other compressors I have can handle hotter signals than the Fatso and this SPL Vitalizer.

Since I work with a lot of this type of music, maybe I would better calibrate the 192s to -12db, don't you think?

Or should I limit the peaks and lower the level on the L2 before the insert instead of limiting the peaks and raising the level.

I hope you have the time to try to understand this post

Thanks again
Old 4th February 2003
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bassmac's Avatar

Why not just select "ALL" your PT faders and lower the overall level of your session? - then put the Fatso on your first insert, followed by the L2.
Old 4th February 2003
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blackcatdigi's Avatar
Ok, so you're making a hyper compressed mix super loud and wondering why that is distorting the next device's input?...

The short version is:

The L2 should be the LAST component in the chain.

But you would do well to run the mix w/o the L2, then import into a new 'maximization' session for 'loudening'...

With regards to your original line of questions, it seems you're chasing your tail...

"Doctor, it hurts when I do this..."
Doctor's reponse: "Stop doing it then."

Try the L2 last. Callibration is not the issue.
Old 5th February 2003
Thank you all for your posts

I was using the L2 to limit some more peaks before going to the Fatso but I was also raising the level although not much.

I tried to calibrate the 192 to -12 but it was impossible. The input would only go to -13.6 :eek: Than I calibrated the inputs to -14. When I tried to calibrate the outbuts it would only go to -14.4 madd so I caibrated it to -15

I'm more happy now with the I/O levels calibrated as it is.

And I'll try to mix with lower levels .
And then apply the L2 in another session.

Old 5th February 2003
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chap's Avatar


I agree with the above post except that calibration will bring your Digi converters in to a range that is more uniform and manageable.

We live in this funny time where music has turned into big, tiring chunks of sound with a total disregard for dynamics. Keep your L2 last, your bit length at 24 as long as possible and make sure when you maximize that you retain some dynamics.
Loudest does not win but will be the first to fatigue a listener..
Old 4th June 2010 | Show parent
Here for the gear

I turned down all the A output screws as far as they would go, and still I get distortion from anything I plug the analog outs into. Am I supposed to have a special $200,000.00 board or something? I almost want to go back to the 003 where I could monitor analog-ly. I'm sticking to optical here.
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