mastering with ableton live

Discuss music production with Ableton Live.
tomperson
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Post by tomperson » Fri Oct 06, 2006 6:09 pm

I tend to work as in the old analogue way, master fader at 0, all the channel faders where they should be to sound good, and no oranges or reds ever. I *think* I've read before that with floating point internal mixing you can indeed very much "overdrive" the master without getting ever at 0 digital (clipping) due to the bigger headroom, but at a price: as you go higher and higher, you start to lose information due to the scaling that goes on when dealing with ever bigger floating point numbers, or something similar.

I think we need Robert here.
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mercyplease
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Post by mercyplease » Fri Oct 06, 2006 6:29 pm

Tarekith wrote:
mercyplease wrote:
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.
I think there's a difference between clipping the master fader, and lowering the final bit resolution.
Hmm, where can we solid answers on this one way or the other?
Can I point out that
This means that you can mix 128 faders at +12dB with the master fader down to -90dB without overdriving the internal mixing buss.
This is true and can I point out that according to digi you wont lose any resolution either. I have to agree because I rendered a test tune in pro tools when they posted exactly how the engine works on the DUC (the white paper) and right enough I couldnt hear a jot of difference with the master fader right down compared with the same mix at unity. Of course I had to normalise the lower fader version. I was impressed to say the least.
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Tarekith
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Post by Tarekith » Fri Oct 06, 2006 7:10 pm

Isn't internal mixing bus PRIOR to any master faders though?

Just out of curiousity, what did you mean by you had to normalize the one with the lower fader? Wouldn't that imply that you were using less bits? Not sure what you meant, that's all.

Cheers for the intriguing and well spoken (ie, no angry people) discussion.

gotascii
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Post by gotascii » Wed Oct 18, 2006 6:26 pm

I'm pretty new to all this and I'm doing my best to follow along but I have a question about clipping the channels. Tell me where I'm mistaken:

Based on http://en.wikipedia.org/wiki/Headroom

I was under the impression that a signal at 0dBFS was full use of your headroom. A signal greater than 0dBFS is clipped/distorted and if Live has 64db of headroom that means it has an alignment level of -64db...right?

If a particular sample causes the channel to register as clipped why would I not have to worry about that? Do the max levels of the channels not really indicate clipping at 0dBFS? Also, lets say I have a sine wave thats clipping my channel, if that is causing clipping inside lives internal mixer which has tons of headroom doesn't that mean when it hits my master level thats only 24bits it would surely be clipping the master?

I must be totally confused but I've sat here for hours reading wikipedia stuff on this and can't seem to wrap my head around it. If this is RTFM stuff can someone point me towards the M? thanks!

DeadlyKungFu
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Post by DeadlyKungFu » Wed Oct 18, 2006 6:47 pm

gotascii wrote:I was under the impression that a signal at 0dBFS was full use of your headroom. A signal greater than 0dBFS is clipped/distorted and if Live has 64db of headroom that means it has an alignment level of -64db...right?
The tricky part is that dBs are relative, 0dB doesn't mean anything unless you know what the reference is.
0dB FS = 0db of full scale, all the bits are '1'.
0dBr = 0dB reference level, whatever the guy writing the data wants to refer to.
0dB in pure audio terms is whatever it says in wikipedia (I forget the number, it's some SPL <sound pressure, IOW audio> level).
Headroom is the difference between full scale and the reference dBr you're using.

Internally in Live there's 60dB (or 64dB) of headroom, meaning 0dBr internally to Live is 60 dB above 0dB at 24 bits. Live uses a dynamic bit allocation technique internally, so it's difficult to say it has any constant dynamic range internally.
If a particular sample causes the channel to register as clipped why would I not have to worry about that? Do the max levels of the channels not really indicate clipping at 0dBFS? Also, lets say I have a sine wave thats clipping my channel, if that is causing clipping inside lives internal mixer which has tons of headroom doesn't that mean when it hits my master level thats only 24bits it would surely be clipping the master?
You've got it except that Live has gobs of headroom inside the code, not at the master out. The internal headroom allows for greater signal processing ability without worrying about clipping internally (as much). The red track level indicators do not represent clipping level, they're a warning, Henke has posted details on this, basically Live gives you wiggle room on the clipping at the output, they're a warning, or a guide, not a hard set level.

To track down clipping internally, watch the meters at the outputs of each plug-in.

I hope that helps, hopefully Henke can chime in on this. You're pretty much got it. In the end, do what sounds the best for you. It's most important on the Master levels.

Deep topic...

dj superflat
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Post by dj superflat » Wed Oct 18, 2006 8:39 pm

i think things like this likely affect how people record/mix in live, and therefore may affect perceived quality of results.

Jan Holm
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Post by Jan Holm » Thu Oct 19, 2006 11:30 pm

Semi new setup here - been trimming all meters and equipment
making my gain structure - and reading up on live.

Kool you can "overdrive" individual channels and then just
trim the master fader.

Import a modern song that peaks about 0 db on every
beat. Gain say +16 dB. Pull the master down by 16 dB.

All kool - until you place a plug on the master. Or did I
miss something here. I think I'll keep the individual tracks
at levels where THE hottest spot in the song hit 0 on the
master.

On the same matter. The mixer meters goes 6 db
above zero. That is just wrong in my head, thats analog
thinking. 0 is max - cant go beoynd that or the file, audio
inface ect will distort. Its kool to have the 6 db gain option
on the fader - but the meters, zero is zero.

Now the manual says - "Live provides this optional visual
feedback for signals that travel beyond 0 dB in any track"

Optional as in flip a button and 0 will max the master meter
:?:

Tarekith
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Post by Tarekith » Thu Oct 19, 2006 11:43 pm

It's generally best to leave the master fader at 0 though.

:lol:

gotascii
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Post by gotascii » Fri Oct 20, 2006 12:57 am

"IE, if you're rendering a 24bit file, and the master fader is set to -48dBFS (just for example), your song will only ever be using at most 18bits (6dB equals 1 bit of resolution in 24bit audio). Even though the final file is still a 24bit file. "

Wait, I did this calculation and I got 16bits? Am I doing somethign wrong:

-48dBFS/6 = 8 and 24 - 8 = 16 so wouldn't you be using at most 16bits?

Tarekith
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Post by Tarekith » Fri Oct 20, 2006 1:41 am

Yeah, I have problems with 2+2 as well. :)

Sorry about that.

Anubis
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Post by Anubis » Fri Oct 20, 2006 3:36 am

Very informative discussion here and no flamers- how refreshing :D
Recently while honing in a mix I would get an occasional (red)spike and I was going nuts trying to correct it by changing levels. Then I isolated the file and hit the Edit button :lol: and was able to remove the offending click using DeClicker in Wavelab. I guess I just wanted to point out that we shouldn't overlook the usefulness of cleaning up the source file if it needs some tweaking.
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dadabot
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thanks a lot for all this info, guys!

Post by dadabot » Sun Oct 28, 2007 12:06 pm

hi all,

i've been using Live for nearly 2 years now.... i've got a painfully acute sense of hearing and i've nearly always been disappointed with the final results of my work... i've encountered many of the issues you've mentioned here and lately, out of despair, i thought to give Ableton a rest coz i just could not get the quality of sound i was expecting from it...

this discussion has been a real eye (and/or ear!) opener for me, so just wanted to say thank you!

got a lot of tweaking to do to some of my mixes so best get on with it now!

cheers!

dadabot
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VST and Plug-in faders going into orange

Post by dadabot » Tue Oct 30, 2007 1:14 pm

hi again,

just wanted to ask a quick question, which has been partially answered in differnt posts within this thread already, but that still bugs me...

i seem to get the same issue with pretty much any VST, plug-in or effect, either Ableton's own or third party, i.e. i use it on one track and the fader for that specific effect, plug-in or VST hits orange/red almost immediately, but the fader for that track does not...

is this comparable with the discussion on the relative importance of track faders going into orange in relation with the master fader, i.e. one of the main subjects of this thread... i mean, should i not be bothered immensely by VST/plug-in/effect faders going into orange as long as the track fader is still within the green? in most cases i find it nearly impossible to lower the vst/plugin/effect values to such a point where they actually do what you want them to do without them going well into the orange, e.g. the perfect example would be an equalizer plugin, where you want it to add some punchy bass to a trak but even at very low levels it's reaching the top of the orange...

i've had loads of problems with distortion lately so i'm trying to get my head around things... i just want my sound to be sharp and crystal clear, but i'm finding it difficult to achieve...

sorry if this has been asked in the past, but any light you could shed on this would be greatly appreciated...


dadabot

SubQ
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Post by SubQ » Tue Oct 30, 2007 1:53 pm

Ok, I'm always following the technical rules for mixing and mastering here, always with satisfatory results, so I can't complain.

But...

try open a finalized album - say, red hot's "stadium arcadium" on a pro audio editor - I use sequoia and wavelab for mastering.

analyse the audio: peak, loudness, etc.

what I've been noticing is that some pro albums always clips on 0db - the red in the peakmeter wont come out for even a sec.

That said, how can we follow the "always put your brickwall limiter at -0,3db to prevent overload in 99,9% of all sound systems whem mastering" rule???

something seems to me a bit obscure these days. In the past, I've always eqing and compressing gently when mix, following the "rules". low Qs, gentle eq adding, all the summing stuff, etc.

After seeing a great dubber mixing putting 12db on some eq frequencies and sounding great, i've started to mix-master less with the eye and more with the ear. My final products has improved, that's all I can say.


so, dispite all the great discussion and knowledge showed here, I dare say:
TRUST YOUR EARS ONLY!! :)

all the best,

coelho
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dadabot
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Post by dadabot » Tue Oct 30, 2007 2:37 pm

thanks a lot coelho!

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