First of all, you may want to differentiate between dBu and dBV because I think you MAY be confusing your +4 and -10 as both being dBu. (I could be wrong, but it's probably a good assumption, since dBV isn't mentioned anywhere. I'm just trying to clarify the situation :wink: ). -2dBu DOES seem an odd level.

Unless you are stating the proper references in your question, anyone can suspect that you are actually referring to -10dBV instead of -10 dBu, but they would be guessing, and the answer may confuse you even more. So, here are some things to consider.

From the following Rane website:

http://rane.com/par-d.html

0 dBu Preferred informal abbreviation for the official dB (0.775 V); a voltage reference point equal to 0.775 Vrms.

+4 dBu Standard pro audio voltage reference level equal to 1.23 Vrms.

0 dBV Preferred informal abbreviation for the official dB (1.0 V); a voltage reference point equal to 1.0 Vrms.

-10 dBV Standard voltage reference level for consumer and some pro audio use (e.g. TASCAM), equal to 0.316 Vrms. (Tip: RCA connectors are a good indicator of units operating at -10 dBV levels.)

Then, once you check which dB reference all your equipment is using, here's a handy little dB conversion calculator (down the page a bit):

http://www.doctorproaudio.com/doctor/calculadores_en.htm

Hopefully, some of this will help answer your question.

Kapt. Krunch

The easiest thing is to take option 1 and enjoy the extra headroom. I would often trade headroom for noise floor, but it does depend somewhat on the make/model of your external ADC.

Another option that you don't mention is to have resistive attenuators on the -2dBu console outputs driving the -10dBV inputs of the ADC and the 002r. Don't forget that -10dBV is roughly -8dBu, so you would need only 6dB balanced attenuators. These are available as in-line XLR-XLR adaptors, or you could make them.

Kapt.Krunch wrote: -2dBu DOES seem an odd level.

- Kapt. Krunch

Allen & Heath does the same thing on some of their consoles. They somewhat arbitrarily about 1/2 of their console outputs as -2dBu and the other half as +4dBu, which is really annoying... Its beyond me.... but I think it has something to do with some vestige of [tape] recorders wanting to deal with -2dBu rather than the hotter 4 signal.... I dont know... Typically the so called "tape outputs" are the -2dBu outputs on some of the A&H's.... which really sucks if your not actually using "tape" (what year is this?).

It might look as though -2dBu is a compromise half-way level between -10dBV (-8dBu) and +4dBu, such that it could be made to work with either at a pinch without too many headroom problems.

In practice, the -2dBu level was probably driven by engineering and marketing considerations. To achieve direct outs and insert outs at +4dBu with a 20dB headroom (i.e. +24dBu clip level), you need at least +/-20V (preferably +/-24V) rails in your channel strip, giving a peak current of around 30mA into 600 Ohm loads. At the time these consoles were designed, this was a tough spec to meet while keeping distortion figures and thermal dissipation under control in a budget product. Halving the voltage from +4dBu to -2dBu, thereby quartering the power, makes a significant difference to the designer. It also put a marketing distance between these consoles and the fully-professional ones where +4dBu into 600 Ohm is standard on all outputs. Labelling the -2dBu outputs as "tape" outs softens the effect of a weaker spec.

Reference levels can be humongously confusing for most. Everybody's example as set above, is both correct and incorrect. How can that be? Because the Japanese have tried to reinvent the wheel their way!

For instance. The Teac/TASCAM 0 DB, -10 DB output reference level, is NOT referenced to .775 volts! Teac referenced their -10 DB output with reference to 1 volt! And so, their reference levels do not correlate to American/European reference levels properly.

As Boswell indicated, I would go for the headroom, especially for popular recording, where signal to noise ratio is not as big a factor as it is with fine arts classical music. Headroom is the difference between retaining the transients or clipping them, as inexpensive equipment so often does.

Head engineer (who's got the pot??)
Ms. Remy Ann David

RemyRAD wrote: The Teac/TASCAM 0 DB, -10 DB output reference level, is NOT referenced to .775 volts! Teac referenced their -10 DB output with reference to 1 volt! And so, their reference levels do not correlate to American/European reference levels properly.

That's why we write it as -10dBV, the "V" standing for 1 Volt. The original 0dBm was the level required to push 1milliwatt into 600 Ohms, which came out as sqrt(600/1000) Volts = 0.7746V r.m.s. When 600 Ohm terminations went out of fashion, the same voltage level was retained, but now into a relatively higher load impedance, so the reference voltage level was re-named 0dBu (u = unterminated). There is approximately 2dB difference between 0dBV and 0dBu, making the difference between -10dBV and +4dBu approximately 12 dB or a factor of 4 in amplitude.

Yes and that's why it's so confusing to most. But the unterminated "dbu", value is in fact generally terminated into at least 10,000 to 50,000 ohms. Just not 600 like the front end of the UREI 1176LN's.

You wouldn't believe how many people I've talked to, that couldn't understand the problem with the underwhelming sound they got, while patching their 1176's, LA3's, LA4's, into their inserts on a Soundcraft, Allen and Heath, Mackie or, any console that wasn't an API, Neve, SSL, Electrodyne, Sphere, Quad Eight, OK MCI also, etc.. And that's because those wonderful old limiters can really only be utilized in a control room that was intended for 600 ohm studio operation and many people here don't know that. And none of those inexpensive consoles inserts can be used with any 600 ohm audio stuff. Of course that's not true for the later non-transformer op-amp balanced input 1176/1178 silver faces, which work fine with your kids console.

I'm not even going to talk about the equipment that is supposed to have a 600 ohm resistor to terminate the output transformer, for a proper and flat response, like my Neves and API stuff. Of which there's lots of.

Unterminated at 51
Ms. Remy Ann David

thanks very much for all of your replys. I am learning so much. Really really great stuff
I realize i did leave out dBV when refrenceing -10. My mistake.

So so far everybody votes setting input of adc to +4dBu. I'm not a big fan of using attenuators on -2 dBu console feeding -10 input on 002 and adc becuse i have a lot of outboard gear that's +4 as well as all my outputs during mixdown. What sort of differences in performance are there between +4dBu and -10dBV? Are attenuators more transparent than transformers?My guess is yes. I havn't thought to much about running everything at -10dBV using attenuators. Any opinions on this?

To be honest I'm a bit worried about the extra noise that will be introduced with the level missmatch. I should have mentioned before that I do a lot of recordings with over 50 tracks and as many as 80 (thanks to quadzillas), compressing the hell out of room mics and such as well as some classical and acustic stuff. 6 dbs or so of noise per track really adds up when you get up there in track count.

Any comments on the transformer option. Am I way off here. Would the transformer option on the direct out of the DDA be able to step up level from -2dBu to +4dBu with the right transformer? Or is the transformer option only for sound perposes?

rambelam wrote:

To be honest I'm a bit worried about the extra noise that will be introduced with the level missmatch.

Any comments on the transformer option. Am I way off here. Would the transformer option on the direct out of the DDA be able to step up level from -2dBu to +4dBu with the right transformer? Or is the transformer option only for sound perposes?

Transformers = potential signal distortion but no thermal noise increase.

Resistor pads = thermal noise increase but no signal distortion.

thanks for clearing that up mises. Mises, when you say "potential distortion" is that to say there are transformers for this aplication that won't effect the signal enough for it to be noticable or negative? If so do you know aproximatly how much will I have to spend for this transormer?

I think i should clarify that the DDA console has an option to add transformers in the mixer itself at the direct out not somthing i would be plugging the dir out into.

I'm at a bit of a stand still not knowing if decent transformers are in my price range or if they're are the way to go.

If someone could fill me in to the performance differences between -10 dBV and +4dBu it would be much appreciated.

soon i will start a new thread with better questions now I have more info

thanks again everyone

rambelam wrote:

thanks for clearing that up mises. Mises, when you say "potential distortion" is that to say there are transformers for this aplication that won't effect the signal enough for it to be noticable or negative? If so do you know aproximatly how much will I have to spend for this transormer?

I think i should clarify that the DDA console has an option to add transformers in the mixer itself at the direct out not somthing i would be plugging the dir out into.

I'm at a bit of a stand still not knowing if decent transformers are in my price range or if they're are the way to go.

Transformer designs for audio, to be honest is not my specialty. I do digital communications theory, not really circuit design... but my knowledge of circuits is probably adequate enough for general advice.

Generally the more desireable transformers for audio are quite expensive (rediculously so... but thats the economics of market value).

I'm not even sure if there are any stock transformers out there specifically designed to convert from the -2dBu level up to +4dBu. If their were... I would buy them and I really wouldnt bother worrying about whether they were astounding quality or not.

Good transformers cost upwards of $120 - $200+ apiece (just from what I've seen)... so you couldn't realistically be putting one of those babies on every single channel on your board... though I suppose you could put it on just a few select channels... depends how many -2dBu outoputs you have.

I would not convert downwards to -10dBV... I'd step up to +4dBu. Just my little opinion on that. If I were going to pursue a route.... thats the way I would go, and I would stick with it... and just buy whatever it takes to make that happen. I think stepping down is not really a great idea unless somebody has a counteropinion on that. Most of your equipment wants to operate in that region anyway... so thats why I say go with that.

Did you mention something about your DDA console has an option for some transformers that you can put on the direct outputs (inside the console)? I would investigate what that costs. Forget about "custom" transformers.... Don't go there unless you want to seriously get into that game, and thats just gonna cost mucho bucks.

Your output is -2 at the insert and/or direct outputs largely because, most of the other devices, they intended for you to patch in, have input level controls. So it's not uncommon to take the insert and/or direct output into a compressor or other outboard device at that level.

Gain staging does not keep everything at the same levels. Levels are dropped and boosted throughout the internal signal path of all consoles, recorders, equalizers, etc.. So I think you should do nothing.

Your theories of the loss of signal to noise ratio on the order of +6 DB over 50 to 80 channels does not hold up. Good engineering is good engineering. We managed to do the same thing, all on analog tape without noise reduction for many, many years, with a signal-to-noise ratio far worse than the electronics in your unbalanced DDA console. So, 24 track wasn't just 24 track as we bounced and transferred many things that made some of our tracks, on the multitrack, over three generations down before the mix even started! Automation, noise gates and riding the gain is how we made our stuff sound good.

So again I say, LESS IS MORE and KEEP IT SIMPLE STUPID. You don't need resistive pads to convert to -10 and you certainly don't need transformers unless your audio seems anemic and needs more iron or, a balanced output? Electronics to boost the -2 output level is more electronics you don't need to feed your signal through. The output transformer option for your console is also not completely necessary unless, of course, as you surmise, they included a "step up transformer". It raises the level, when the windings on the secondary are greater than the windings on the primary side (which can also be run in the opposite direction).

API consoles all had transformer outputs that included 3 separate secondary windings. So you had a turns ratio of 1: 1 for a +20 DBm output or, 1: 2 for a +24 DBm output for, 1: 3 or a +28 DBm output drive capability. The other advantage is, as in my API 3124m units utilizes 2 separate isolated outputs from the same transformer. Great to use as a 1 x 2 active splitter or dual isolated outputs.

While Transformers can and do introduce distortion, the largest problem is the low-frequency distortion and the saturation of the core, that occurs with cheap Transformers. Good transformers can actually benefit your audio and/or color it in a pleasing way. Transformers are either specified as low-level microphone or high-level line devices. You wouldn't want to use a line transformer for microphone purposes and vice versa.

If transformers were bad, Neve consoles would never have cut a hit much less a demo. My audio through my Neves go through from 8 to 10 transformers from microphone input to bus output, except in the case of the individual modules which will only force you through 2. And you never heard anybody complain about that.

Tranny heaven
Ms. Remy Ann David

If somebody offered me 6dB more headroom at no penalty for most of the type of material I was recording I'd bite his hand off. It's one of the reasons that engineers run the gain trims lower and the faders higher on consoles with limited headroom.

So I would go with driving the +4dBu inputs of the ADC or 002r. You just might consider having a few inline XLR 6dB attenuators in your bag for the rare occasion that you come across very quiet material and need the extra dynamic range. That would be cheaper than splashing out on high-grade 1:2 transformers that you would scarcely ever use.

By the way, the thermal noise of a 600 Ohm resistor is about 129dB below +4dBu over a 20KHz bandwidth, so is not a factor.

Boswell wrote: If somebody offered me 6dB more headroom at no penalty for most of the type of material I was recording I'd bite his hand off. It's one of the reasons that engineers run the gain trims lower and the faders higher on consoles with limited headroom.

I LOVE all you 'real' engineer types. This is great stuff here. GAFF STICKY THIS!

This Boswell quote I have hi-lighted is a key to gain staging for all you guys and gals with the Mackie/Allen Heath/ Soundcraft(Ghost and below) type of consoles.

I get what I need on my Ghost exactly like this though I do regulate the input voltage at a steady 128 vac. Its within specs so I dont worry about damage per se. The rail voltage comes up a touch and the pres get creamy and punchy while the EQ has an increased bite and the sweep is much more effective.Even surgical if need be.

I gotta agree with Boswell on this entire point. Take the headroom. Buy some outboard(or build) attenuation for needs...Like Remy....dont do nuttin. Your outboard likely has attenuation built into the front ends and makeup gains going out. While you might have to be more astute in your balancing act with this setup, I see no reason it should cause problems in recording materials and certainly with high track counts the added headroom would be to your advantage.

Did I mention that I LOVE you engineer folks?

Davedog wrote:

I LOVE all you 'real' engineer types. This is great stuff here. GAFF STICKY THIS!

Did I mention that I LOVE you engineer folks?

Now what fun would it be, being an [electrical] engineer, if we can't nitpick at little things like adding 449.7 nanoVolts of thermal noise (making the cumulative AWGN a whopping (-116) to (-122) below +4dBu.

Somebody has to worry about a few nanoVolts or microVolts of noise 8)

Thank God I became a mathematician. Dimensionless variables. All constants set to one. Life is good - at least in my day job.

Awwwwwwwww That's sweet Dave. I love engineers too, especially good ones. Console size does matter. After all, I've had the best.

Throw that Dog a Microbone or maybe a Dogphone?
The Demure Ms. Remy Ann David

Mises wrote: [quote=rambelam]

Good transformers cost upwards of $120 - $200+ apiece (just from what I've seen)... so you couldn't realistically be putting one of those babies on every single channel on your board... though I suppose you could put it on just a few select channels... depends how many -2dBu outoputs you have.

Holy Cow! You looking in the wrong spot. We get $90 Jensens or $ 75 Cinemags without even trying.

Sorry have to bring the geek out on this one.
Check out:
http://www.sengpielaudio.com/calculator-db-volt.htm

Heres some numbers:
-2dBu = 1.74Vpp
+4dBu = 3.47Vpp

-10dBV = 0.89Vpp

You need a gain of 2 to go from -2dBu to +4dBu
or you need a gain of 1/2 to go from -2dBu to -10dBV

Are these short unbalanced lines?
If yes a resitive divider will work just fine provide you pay attention to the input impedence of the -10dBV inputs.

Or use a simple class A fet amp with a gain of 2, If you would rather go up +4dBu.

I can send you circuits for either solution if you like.

The class A fet design is cheap compared to xfmrs. Although I do love cinemag! I use them in all my Apex 460 mods.

Best of luck.

The actual reason why the operating level in many british mixers (and subsequently many japanese and US manufacturers) is -2dBu is that circuitry using commonly available IC's provides an actual clipping point of ca. +20dBu. Running the board at +4dBu internally would provide only 16dB of headroom, which was deemed insufficient, which is true in live applications, and that permeated to studio mixers. That's why they chose to run all internal signals at -2 dBu and interface with the external world via balancing/debalancing stages providing 6 dB gain/attenuation. The commonly used EBOS (Electronically Balanced Output Stage) take advantage of 6 dB increase of output capablity (typically + 26 dBu into a balanced load). Some high-end mixers used balanced inserts running at +4, but most of the products had unbalanced inserts at -2, in order to save circuitry.
Anyway, most of the outboard gear can make up for the difference. On a 1176, you just crank up the input and reduce output accordingly, on a 160, you just set the threshold 6 dB lower. For EQ, it shouldn't matter, except if it's very noisy.
In most cases, it will be transparent to the user; that was not the case when Tascam ran their mixers at -10 dBv (-7.8 dBu) from beginning to end.
You may have difficulties with a Fairchild 660 or 670, just as you would with a mixer that runs its inserts at +4, because the 660 was designed for 600 ohms input source AND 600 ohms output load (impeadance-matching), which no modern gear supports.
Most equipment nowadays is designed for impedance-bridging, i.e. low output impedance (10-100 ohms) and medium-high input impedance (5-50 kohms).

Ummm, you really can't run a classic 1176 on the inserts of most modern-day mixer boards. The originals had 600 ohm terminated balanced bridging inputs. Now all the latter models which were silver faced utilized a 10,000 ohm, balanced integrated circuit chip, transformer less input. Those you can patch on inserts of most budget consoles.

And Teac/TASCAM referenced their -10 DB standard at .3 volts as it relates to a 1 volt 0 DB reference, as opposed to a .773 volts, 0 DBm into 600 ohms for a + 4 DB zero reference, which is 1.25 volts RMS. If that isn't enough to confuse you? I don't know what is? Yeah, most integrated circuit chip outputs can't deliver more than +18 DBu, into 10,000 ohm inputs. And so for most folks who don't have the pleasure of using a fully professional high-end console, you're stuck at much less headroom. Headroom is the difference between professional sound & beginner sound. But beginners can sound like professionals if you make your self some headroom.

Big mouth +28dbm broad, yeah baby.
Ms. Remy Ann David

RemyRAD wrote: ... Big mouth +28dbm broad, yeah baby...

yeah baby

an end of post quip this maybe
but
to understand it
is to understand the whole point

to use and interface old and retro gear with modern mixers and digital interfaces etc
and
to get them into the sweet spot

you need to understand why and how Remy is a
" big mouth +28dbm broad "

funny yes
but this is no joke

RemyRAD wrote: Ummm, you really can't run a classic 1176 on the inserts of most modern-day mixer boards. The originals had 600 ohm terminated balanced bridging inputs. Now all the latter models which were silver faced utilized a 10,000 ohm, balanced integrated circuit chip, transformer less input. Those you can patch on inserts of most budget consoles.

I can only disagree with that. Most of these "budget consoles" have the capability to deliver +18dBu into 600 ohms, which is plenty even for a "classic" 1176, unless you want 30dB+ compression.

well I disagree

and it comes down to how much headroom you have chosen to have in those "budget consoles" OR the modern DAW

very difficult to get typical recorded program up into the sweet spot of things like transformered 1176 and LA2s

I have been asked a few times to mod these units to add just a little gain at the front end to suit the insert sends ... (or add a line driver to the send)
...
then PAD the output back a little to bring back into the insert return

I do think it is at the output that brings much of the sound people are looking for
so terminating and the PAD is probably the first thing to try.
The Gain Make Up Amp in these units often have enough to get you going but the risk is that you may get some noise with it. That's why finding 10db (or a little more) of signal at the front can help.

Kev wrote: well I disagree and it comes down to how much headroom you have chosen to have in those "budget consoles" OR the modern DAW
very difficult to get typical recorded program up into the sweet spot of things like transformered 1176 and LA2s

It is your absolute right to disagree, but I know for sure that an 1176 is not expecting more than 0dBu to operate properly. When I installed the cutting room at Barclay Studios in 1976, we had a Neumann mixer, which ran on a single 24 V rail and used LF356's, hence offering only +16dBu output capability on the unbalanced insert points. The 1176's (with xformer inputs and 600 ohms constant-z attenuators) had enough gain to cope with that.

I do think it is at the output that brings much of the sound people are looking for so terminating and the PAD is probably the first thing to try.

I agree with that; The output xformer in the 1176 wants to see a 600 ohm impedance because it has a relatively high leakage inductance that makes it ring if not properly damped.

The Gain Make Up Amp in these units often have enough to get you going but the risk is that you may get some noise with it.

No, because the input attenuator is before the input preamp. Having less attenuation before the input preamp doesn't change noise.

no no

I was referring to signals that might be for you, a very low signa,l with a great deal of headroom
and could be much lower than 0dbu

and I can see how a single 24v rail with around +16dbu unbal would also be quite low for a max signal

some people seem to want to see 20db+ of headroom

as for the gain noise / issue
to CRANK the gain make up amp can make noise for within the 1176

remember there are 1176's out there that were NOT the LN models
Low Noise

and I was referring to these things genericaly and including LA2's .... these things can be living being that have lived hard lives
and it is hard to find two the same

and for things like the Pendulum compressor +16dbu unbalanced would not get it into it's sweet spot and a PAD would be needed to bring it back into one of these desks

You seem to know enough about your gear to make your own choices and the comments here from Remy and I are not really directed at you.

IF all of this gets people thinking about how they interface there gear then this is a good thing.

While we are here on the 1176 ... the manual does give some good notes on how to interface it
.... read the manual

you can find PDF's on the net ... my site might even have one

Most of this confusion could be solved if the audio industry just adopted what the rest of the world uses for a voltage reference:

1 Vrms sine wave = 0 dBV

there's just something refreshing about 20 log (1) - ya' know?

hi folks,is nice to be in your group,isa class

dpd wrote: ... what the rest of the world uses for a voltage reference: ... - ya' know?

too much history for that

Video is a 1Vpp world with 75ohm
but now with digital that doesn't seem so important anymore

the RF world has it's own issues

The 1Vrms has no regard for load
1 mW onto 600 ohms ... because that was the standard = .... you can google that

things change
digital interfacing had the chance to solve level issues as all is referred to 0dbFS
but
now we have clock issues

it's all fun

this is something i too have a hard time getting through my head.
on my mix wizzard,the inserts are -2.
however on my new zed r-16, the inserts are 0 DBU nominal to +21 DBU max.
i run my outboard comps(insert) at + 4. is this correct?
i tried them at -10 DBV and i get a hotter signal.
it works both ways, wich is best?

boxcar, post: 298770 wrote: this is something i too have a hard time getting through my head.
on my mix wizzard,the inserts are -2.
however on my new zed r-16, the inserts are 0 DBU nominal to +21 DBU max.
i run my outboard comps(insert) at + 4. is this correct?
i tried them at -10 DBV and i get a hotter signal.
it works both ways, wich is best?

You should run at +4 if you want to avoid clipping or saturation at the input of the compressor. You will need more input gain on the compressor than if you had real +4dBu signals, and correspondingly less make-up gain at the output of the compressor.

You will indeed get a hotter signal if the compressor is set to -10dBV, but there is about 12dB less headroom for your signals at that setting, so you are in danger of saturation on transients before the compressor has the chance to get to work on them.

Say, for example, the compressor input headroom is 18dB. This implies input saturation levels of +22dBu and +8dBV (~+10dBu) at the +4dBu and -10dBV settings respectively. Using your Zed-R16 to feed in a level of 0dBu nominal, +21dBu peak would result in the signal being handled correctly with the compressor set to +4dBu, but at the -10dBV setting would cause the top 11dB of peaks in the signal to be clipped due to saturation.

Boswell, post: 298821 wrote: You should run at +4 if you want to avoid clipping or saturation at the input of the compressor. You will need more input gain on the compressor than if you had real +4dBu signals, and correspondingly less make-up gain at the output of the compressor.

You will indeed get a hotter signal if the compressor is set to -10dBV, but there is about 12dB less headroom for your signals at that setting, so you are in danger of saturation on transients before the compressor has the chance to get to work on them.

Say, for example, the compressor input headroom is 18dB. This implies input saturation levels of +22dBu and +8dBV (~+10dBu) at the +4dBu and -10dBV settings respectively. Using your Zed-R16 to feed in a level of 0dBu nominal, +21dBu peak would result in the signal being handled correctly with the compressor set to +4dBu, but at the -10dBV setting would cause the top 11dB of peaks in the signal to be clipped due to saturation.

thanks for that.
it's good to know i've been doing it correctly but i didn't know why until now.