Skip to main content

This is a long-winded help request for calibrating my Yamaha 01v96 mixing desk for the purposes of mixing, live work and recording.

I read a brief description on tweakheadz but it fell short of telling me exactly how to do it. I feel this is a very important point so I need and would be greatly thankful for good descriptive help / advice on doing it.

If I describe what I have gleaned from the article so far along with the gaps I need to fill to realistically accomplish this.

1. Set all faders to 0db - this is fine and I presume it means all channels and master too.
2. Find a reference tone of 0db - this is fine as the 01v96 has the ability to generate the reference tone.
3. Adjust the trim till the level meters reach 0db - there is a lot of information missing here. I need to know the following:

a) Where the reference signal goes and how - I am presuming a physical patch need to be made from an output of the desk and into all the individual channel input preamps.
b) How to adjust the gain - I am presuming that the trim is the adjustment of the gain pot on the channel input until the level meter peaks at 0db.
c) Output levels of instruments that go into the desk – Ok so we have got a reference signal of 0db into the desk and calibrated each channel to 0db. How does this actually work with all the different instruments that get plugged into the desk. For example – I have a guitar pod that pumps out I believe +4db and an organ sound module that gives out –10db and then there are vocals at different levels etc. How does all this reference back to out 0db setting – surely +4db will be too much and –10db to little and so the calibration is point less.

Am I missing the point?

Please help with the above problems,

Ta,

Alex.

Topic Tags

Comments

Zilla Thu, 08/11/2005 - 09:59

alfonce wrote: Am I missing the point?

I think so.

One does not need to calibrate a console like a tape machine or a/d. As you already mentioned, consoles will be fed by multiple sources with varying signal levels. You need to tailor the gain structure of each source/console channel individually. The point is to adjust the gain of each device in your signal path to maximize its signal-to-noise ratio while minimizing distortion (assuming clean sound is the goal.)

anonymous Thu, 08/11/2005 - 10:34

0db

Hi,

well I really do wan to calibrate in some way as I'm ghaving mix problems and I want to sort it before I go any further.

The pods kick out +4db for example - how do I calibrate these so they never overload the desk and the sound module kicks out -10db again how do I calibrate soits not under but never over either???

The 0db ref tone sounds good but I cant see how it can be utilised realistically,

please help

ta
Alex

Zilla Thu, 08/11/2005 - 11:29

Re: 0db

alfonce wrote: ...I'm having mix problems...

What do you mean? Does this mean you can't balance sounds properly?

alfonce wrote: how do I calibrate so its not under but never over either???

You must trim each sound individually on each channel for each individual session. It does not matter that the Pod nominal operating level is +4dBu. What does matter is what the actual signal's level is in that particular situation. Play the sound and trim the gain so that it doesn't overload any section of the channel strip, but so that the signal is still robust. Use the console's meters to aid you in this.

alfonce wrote: The 0db ref tone sounds good but I cant see how it can be utilized realistically

Not for this situation. As I said before, one does not calibrate a console. It is not a set-it-and-leave-it kind of deal. The engineer must tweak for every situation's unique set of signal levels.

Hope this helps.

KurtFoster Thu, 08/11/2005 - 14:36

This is all about internal gain stageing ... what we call gain managment and really has nothing to do with calibration.

Each channel on the 01V has a solo button. In solo you will be able to see that channels levles on the meters without any other channels. Solo the first channel with the fader set to its nominal setting (usually marked "0" on a mixer about 3/4ths the way up its travel) and set the input trim so the meter reads about -6dB. Do this for all channel sources and you should be ready to mix.

BTW, this is the most basic of operations that need to be done when recording and mixing audio, much like the proverbial "babies first step". If you don't know how to do this you shouldn't be recording and you absolutely have NO BUSINESS mixing a live show. :shock: I know that sound harsh, but it's real ....

Find someone who knows what they are doing and beg borrow steal, whatever you need to do to get them to teach you the basics.

anonymous Fri, 08/12/2005 - 00:18

Thanks

Hi there and thanks,

well in the past I have always set levels according to their loudest possible in the track - in my case its my whole live set.

Last night I went and did this again and set the drums to peak at
-3db. You mentioned the master peak level meters. These will only give tyou the soloed track reading acurately if you have the master fader set to the top - as in 0db.

I guess when I tread that article it made me jump cus I never callibrated in that way vefore and I thought I maybe have made a mess from the ground up.

However, Like I said I have always set the channels according to the loudest possible scenario occouring down that channel and it has seemed to work so far.

I think my issue is the cumilative gain effect - this is making a slight(inaudible as the 01v has a lot of headroom before clipping +24db I think)overload internally even though the master strip still shows a cumilative peak of -3 db. So what I am going to have to do is re set eveerything and just bring it all down a little which aint a small job as I have 15 songs based wiith an mpc4000 at the heart which in turn has its own mix capability. The mpc fires into the desk via adat at 0db I'm presuming so I will have to remix the mpc stuff first and then see how my live instruments - guitars, drums, organ etc fit into this scenario.

Tat being said I have always used my drum track as my yardstick to mixing as a first port of call and move from there.

hmm

KurtFoster Fri, 08/12/2005 - 09:19

The general consensus is that shooting for -6dB on tracks and the 2-bus for the loudest parts, allows enough headroom for any unforeseen peaks that may occur. You do not want to have any overs .... ever :!:

Different mixers show 0dB at different points on the faders. Some mixers have 10 dB of extra gain built in to each channel and some don't. Regardless if it's 3/4ths the way or all the way open .... set the channel to "0" and adjust the input attenuator trim.

One way to tell if the gain staging is correct .... ( I learned this one from Roger Nicholas) is to look at the mixer after a mix is set up ... are the faders all pretty close to the 0 point? Are all the faders set in a reasonably straight row, or are some of the set very high and others very low? A mix that was recorded correctly and the gain staged correctly will usually have all the faders set to near the same place across the console ...

anonymous Sun, 08/14/2005 - 09:16

with or without

Hi there,

well thanks a lot for that - I'll set all faders inluding the master to 0db(that means the master will be as far up as it can go as 0db on the master is right at the top of the fade strip). Then I'll adjust the makeup gains to shoot for about -6db on the level meters. Do I then pull the master back to about -6db setting, thus lowering the overall by -6db again?

What would you suggest with my adat inputs from my mpc4000?
There is no gain stage here as its digital. Am I to adjust the faders so the loud parts are peaking about -6db or am I to adjust them relative to say the drums and bass which are analog and do have gain stages??

Whats your opinion regards recording to disk - with or without eq and or compression. I already know recording with fx is a no no unless your 100% sure and cannot achieve the sound after the fact etc.

I read somwhere that you should record with just a limiter - leave all else till mixing. Thle limiter should be set to say -1db on the threshhold with a ratio of infinity - hence nothing but nothing gets above -1db. So inputs are still commuing in way below -1db so the limiting isnt really there to shape the sound bt just to catch any in audible blips.

Thanks again,

Alex