any mixtips? My master keeps peaking on half volume

Discuss music production with Ableton Live.
dj mush
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any mixtips? My master keeps peaking on half volume

Post by dj mush » Fri Jun 20, 2008 7:15 pm

hi,


i'm having some problems concerning mixing my tracks in ableton.

i work in session view and have 4 songs in one liveset.

the problem i have is that my master channel allready turns in red even when it is only half open....
i assume my problem is that one of the tracks is too loud, but when i isolate tracks, then none of them turns red.

also if i lower volume of the drums for example, then the punch dissappears obviously..

does somebody have some tips concerning this?

-having a good output on the master (so when i have to play somewhere that i plug in the laptop and have a strong signal coming out)
without turning red
-preserving punch in drums

i tought that the master should be on the little horizontal dash (0)
but when i do that its completely in red (as i mentioned, its allready red halfway)
are my sounds and samples too loud?
do i have to drop the volume on all my wavfiles in this set?

thank you!
ADKproAudio, MPC 2000XL, Ableton, Juno 106, MOTU 828mk2, Gameboy, Macbeth M3X2, Acces Virus Polar, Moog Little Phatty II, Akai mpd 32
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Tone Deft
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Post by Tone Deft » Fri Jun 20, 2008 7:21 pm

it's the quantity of tracks overloading the summing bus. this is a great post

http://www.ableton.com/forum/viewtopic. ... 429#659429
Tarekith wrote:Cue my favorite Roger Nichols article:

"I heard the mixing buss in Pro Tools is no good. Everyone says I should mix through an external analogue summing buss."

Someone asked Al Schmitt how he mixed a record. Al answered "I just turn the knobs until it sounds good." You can't argue with that.

The first thing I had to learn about audio engineering was signal flow. You have to know how to get the audio signal from the microphone to the recorder and back to the speakers so you can hear what you are doing. The second item is gain structure. At any point in the signal path you have to keep the signal higher than the noise floor but lower than the point of clipping and distortion. Everything else is going to be easy. Just twist the knobs.

Every console is designed to add signals together before they come out as finished mix. It is called a mix because the individual tracks are mixed together. I rather fancy the English term 'two-track reduction': it is being reduced from 24 tracks to two tracks.

Physical consoles usually have a stereo mix buss 'summing amp' for each group of eight modules. These summing amps then feed another summing amp connected to the master output module. During the mixing process the master fader is turned all the way up. As individual tracks are turned up their audio is heard through the master fader and the level is registered on the main output meters. As more and more audio channels are introduced to the mix, the overall master level starts getting high, so the master fader is turned down a little to compensate. This work flow proceeds in a loop until the mix is getting pretty far along.

At some point the engineer looks at the gain structure of the mix he has going on the console. He has learned that by running the individual faders high and pulling down the master fader he runs the risk of overdriving the summing amps with too much level and adding distortion to the final mix. The engineer will trim down all of the track faders by 6dB or some similar amount so that the master fader can be brought back up to zero.

This method has worked for decades to keep the audio quality as high as possible while remaining within the limits of the console's design, but for some reason engineers ignore this procedure when mixing inside a DAW (digital audio workstation). When asked why they don't perform this requisite task the answer is always "It's digital, you don't have to do that." All of the 78 track faders are up near zero and the master fader by now is down to -40dB. Soon the engineer starts to complain about how gritty and distorted digital sounds.

How do they fix it? They connect the DAW to a console. At the console they either trim down the inputs or pull down the track faders to prevent the summing amps from clipping, and they make sure that the master fader is all the way up. "Hear how much better the mix sounds through a console?"

Sound familiar? I know all of you have run across this situation from one end or the other. The smart guys who saw this wanton disregard for gain structure quickly designed 'outboard analogue summing boxes', charged a lot of money (because it can't be good if it doesn't cost enough), and made a fortune. Good for them. Too bad I didn't think of it.

Because Pro Tools was the most visible professional DAW, Digidesign took the brunt of the criticism. "Man, I can't mix inside Pro Tools, their internal mixer sucks." Although there were tons of good-sounding records made and mixed in Pro Tools by engineers who knew how to turn down a fader, the majority of the forums on the Web hosted tons of complaints. "It shouldn't do that, it's digital."

Digidesign have updated their internal mixer to 48-bit. This means that you can mix 128 faders at +12dB with the master fader down to -90dB without overdriving the internal mixing buss. There will not be much room for a final fade, but at least Pro Tools is now being idiot-proofed. Me, I prefer to watch what I am doing and trim all of my faders down so that my master fader stays at zero. It has worked for me since the '60s and continues to work for me in whatever digital DAW I mix in.

PS: Just so you know, I do tell my clients that "I only mix on dual 64-bit processors with a 48-bit mixer fed by 16 gigabytes of memory and an on-line RAID5 disk array of 8 terabytes and a 15 gigabit-per-second fibre-optic Internet connection. You do hear the difference, don't you?"
which is why Live needs a better bussing system, track groups and the ability to adjust track faders as groups.

you need to go through and turn down all your tracks, remix them at a lower level.
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dj mush
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Post by dj mush » Fri Jun 20, 2008 7:24 pm

thank you!
:D
ADKproAudio, MPC 2000XL, Ableton, Juno 106, MOTU 828mk2, Gameboy, Macbeth M3X2, Acces Virus Polar, Moog Little Phatty II, Akai mpd 32
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dj mush
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Post by dj mush » Fri Jun 20, 2008 7:48 pm

is it normal that it allready clips when i have only 3 clips playing...?

a few wav clips and a drumrack.

the wavs are - 8db and so is the drums on the drumrack...
do I have to pull them even more down?
is that normal?

or should I first mix the row in session view where the most clips are situated (for example row 20) and if that doesn't turn red, mixing all the other clips and rows according to that one row
(if you understand what i mean)
ADKproAudio, MPC 2000XL, Ableton, Juno 106, MOTU 828mk2, Gameboy, Macbeth M3X2, Acces Virus Polar, Moog Little Phatty II, Akai mpd 32
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Tone Deft
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Post by Tone Deft » Fri Jun 20, 2008 7:54 pm

'normal' is a difficult term, there's a billion songs out there, I have no clue which one you're making.

it is what it is, if you understand the article above (it was new to me when Tarekith posted it) then just use your ears.

if you have a heavy, deep thumpy kick drum going that could be the culprit.

dunno, I'm not much of a producer/mastering guy, I'm more of a player than a recorder. someone will chime in with better advice.

btw Live doesn't use the term 'rows', tracks are the vertical things, scenes are horizontal. the boxes are called clips. good problem description though.
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dj mush
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Post by dj mush » Fri Jun 20, 2008 7:59 pm

thanks again!
i'll follow my ears then i guess :-)

cheers!
ADKproAudio, MPC 2000XL, Ableton, Juno 106, MOTU 828mk2, Gameboy, Macbeth M3X2, Acces Virus Polar, Moog Little Phatty II, Akai mpd 32
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Post by djsynchro » Fri Jun 20, 2008 10:30 pm

This is bollocks! Pro Tools is not a floating point system.

There is no gain staging in a floating point system, you can either turn the master fader down or slap a utility in the master channel and trim it down... that's one of the things it was made for!

You don't have to turn down your other channels, watch when they peak over zero... it's still green meaning, it's past 0db, but NOT clipping, (RTFM) you can only clip the audio at the master fader, just turn it down.

And there's a good reason to have your faders high, they have more resolution at the top, all faders analogue and digital have a logarithmic audio "taper" to match how your ear works, the resolution at the top is greater.

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Post by Tarekith » Fri Jun 20, 2008 11:23 pm


Tone Deft
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Post by Tone Deft » Fri Jun 20, 2008 11:25 pm

Tarekith - was a I wrong, does synchro make a point? I'm still learning this stuff.

TIA.
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Post by djsynchro » Sat Jun 21, 2008 12:05 am

Look Pro Tools (TDM that is, RTAS is also floating point) is fixed-point math.
It's supposed to be theoretically sonically superior, but the deal is, you do have to watch your levels, to make sure you use all the available bits, and also to make sure things don't clip.

With 32-bit float, the audio is always 24-bits, and the 8 remaning bits are used to scale the volume. The advantage is that for all practical purposes such as mixing etc. it doesn't really matter if you have a wimpy signal here or a meter peaking past 0db there.

It's only at the master fader that the audio is 24 bit linear again so there you have to get a nice hot level, without going over 0db. That's why the peak indicator turns red there when you've had a clip.

Post by his holyness Bob Katz here http://www.digido.com/index.php?Itemid= ... e=articles

Tone Deft
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Post by Tone Deft » Sat Jun 21, 2008 12:10 am

djsynchro wrote:Look Pro Tools (TDM that is, RTAS is also floating point) is fixed-point math.
It's supposed to be theoretically sonically superior, but the deal is, you do have to watch your levels, to make sure you use all the available bits, and also to make sure things don't clip.

With 32-bit float, the audio is always 24-bits, and the 8 remaning bits are used to scale the volume. The advantage is that for all practical purposes such as mixing etc. it doesn't really matter if you have a wimpy signal here or a meter peaking past 0db there.

It's only at the master fader that the audio is 24 bit linear again so there you have to get a nice hot level, without going over 0db. That's why the peak indicator turns red there when you've had a clip.

Post by his holyness Bob Katz here http://www.digido.com/index.php?Itemid= ... e=articles
I'm not disagreeing with you, but I'm not reading you addressing the notion of bus summing, which is what I believe the problem is. I understand that you can't really clip at the track, it's the summing bus from the tracks to the master.

I guess the question would be, what argument would you make to dismiss the bus summing idea?

Tarekith didn't really answer anything. :(

why bother mentioning Pro Tools?? fuck Pro Tools! ;) :P
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Post by Tarekith » Sat Jun 21, 2008 12:54 am

he's right, its almost impossible to clip the summing engine or buss as you call it. But you can clip the master as mentioned.

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Post by stringtapper » Sat Jun 21, 2008 12:59 am

Tarekith wrote:he's right, its almost impossible to clip the summing engine or buss as you call it. But you can clip the master as mentioned.
Then what causes the distortion that the first article talks about? The part he talked about where the master fader is way down and all the tracks are at 0db.
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Tone Deft
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Post by Tone Deft » Sat Jun 21, 2008 1:00 am

cool, thanks, I've been doing it wrong. :oops: I rarely go over 8-12 tracks though, and I suck.

props Dsjynchro. :D
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Post by Tarekith » Sat Jun 21, 2008 3:41 am

stringtapper wrote:
Tarekith wrote:he's right, its almost impossible to clip the summing engine or buss as you call it. But you can clip the master as mentioned.
Then what causes the distortion that the first article talks about? The part he talked about where the master fader is way down and all the tracks are at 0db.
Pro Tool is not a floating point application, especially with the earlier versions. In that case, it WAS/IS possible to clip the channel faders, unlike in Live.

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