mute everything except for your kick. now lower/raise the kick level until it gets to around -6db on your main output level meter . now unmute your bass and lower/raise the level of the bass until the main output level goes up from the -6db to about -4.5db or there abouts. now bring all other parts in until you are hitting 0.0db or very slightly under.

now render and listen. do the tweeking that needs doing and work from there.

this is what i was told and i swear by this now.

hope this helps

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quote:

Originally posted by richg101 mute everything except for your kick. now lower/raise the kick level until it gets to around -6db on your main output level meter . now unmute your bass and lower/raise the level of the bass until the main output level goes up from the -6db to about -4.5db or there abouts. now bring all other parts in until you are hitting 0.0db or very slightly under.

Not a bad advice, just remember that all frequencies take up your headroom, so remember to high pass around 35-40 on all tracks. Otherwise you might go up to -6 dbfs without actually "having" the sound and the mix builds up to 0 dbfs too quickly... if you know what I mean.

Oh - and just to be on the safe side. We are assuming that you're mixing on Cubase, Logic or another 32 bit floating point based mix system. In the case you're not (i.e. if you're mixing on Pro Tools or any other 24 bit fixed point engine), you should disregard the advice about going to -6 db etc. You need a lot more headroom on fixed point bit systems to avoid hitting your master bus to hard and thus clipping the mix.

Regards

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RiValMan

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I don't like to mess with the individual faders instead i bring the master fader down until its not clipping anymore..I don't adjust each individual fader since your ears get used to the sound you can mess up a mix by 'fader following' & when you hear the track the next day with fresh ears you're in for a surprise..That's why i mix everything according to what sounds good to my ears & then if its clipping i just pull down the master..

What i'd like to know though is whether i'm losing quality if i put down the master fader by say 20 db or so?

I'm using Cubase SX 3 & i bounce down in 32 bit..

with regards to losing quality..

i would have thought that needing to lower main output by 20db might suggest that parts might be peaking on your individual channels. however i am not 100% sure the 0db rule counts for the individual channels.

im sure there is a perfect level you should use. personally i would set the master to the click point and work from there.

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quote:

Originally posted by s-cube I don't like to mess with the individual faders instead i bring the master fader down until its not clipping anymore..I don't adjust each individual fader since your ears get used to the sound you can mess up a mix by 'fader following' & when you hear the track the next day with fresh ears you're in for a surprise..That's why i mix everything according to what sounds good to my ears & then if its clipping i just pull down the master..

Yes, but you can only do this because you're mixing in a 32-bit floating point environment (doesn't matter if you mix in 16, 24 og 32 bit btw. The internal bus in i.e. Cubase will always mix in 32 bit floating point regardless).
If you're working on Pro Tools (or another 24 bit fixed point system), you cannot just turn down the master. The internal signal path has already clipped the signal.
Please note that if you use DSP cards (UAD-1, Powercore, SSL Duende etc.) you should also be aware that they might work with fixed point processing.

quote:

What i'd like to know though is whether i'm losing quality if i put down the master fader by say 20 db or so?

Well, you're not clipping the signal as long as you're in Cubase or Logic (don't know if FL is 32-bit floating), but you might still "loose" some quality if you're not fully using your bit depth. When working to a 16 bit setup you don't want to lower the output volume too much as you might hear bit quantization noise and thus have a lower quality.
But if you're working with 24 bit (or even 32 bit) files, it doesn't matter if you don't use all your "bits" available. You already have a scale that goes way beyond what is human audible. So my answer would be: No, you're not losing quality.

Regards

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RiValMan

- Pumping the bits using:
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I work in 24 bit & use Cubase SX 3..I'm still confused about this digital clipping issue so i decided to do a test..I made sure one of my tracks was clipping by a couple of db's & exported it at 32 bit floating point (using SX 3)..Then i put that through an L2 with 3db of gain reduction in sound forge..

Now here's the weird thing if i do a 'detect only audibly noticable clips' or 'detect all 0db clipping' on the wav file in sound forge it show that there is no clipping..However if i do the same thing on the mp3 file it detects the clipping..

I uploaded a short sample of the file so you can hear for yourself..Can you hear any clipping?!

http://download.yousendit.com/32952C37672232C8

quote:

Originally posted by s-cube I work in 24 bit & use Cubase SX 3..I'm still confused about this digital clipping issue so i decided to do a test..I made sure one of my tracks was clipping by a couple of db's & exported it at 32 bit floating point (using SX 3)..Then i put that through an L2 with 3db of gain reduction in sound forge.. Now here's the weird thing if i do a 'detect only audibly noticable clips' or 'detect all 0db clipping' on the wav file in sound forge it show that there is no clipping..However if i do the same thing on the mp3 file it detects the clipping.. I uploaded a short sample of the file so you can hear for yourself..Can you hear any clipping?! http://download.yousendit.com/32952C37672232C8

You have to distinguish between the mixing process and the bouncing process.

You can NOT clip a signal when mixing in Cubase (well, in theory you can, but not in real life). The reason is that all calculations are done using floating point. You can, however, clip a signal in a plugin if the plugin uses fixed point processing, but I would say that it's unlikely.

TDM and most DSP systems use fixed point. I don't know if there are any native plugins out there using fixed point processing. I know there are people out there who have been trying for years to get a statement from Waves whether or not their plugins (or which plugins) use 32 bit floating or 24 bit (or even 48 bit) fixed point. But it remains a mystery.

Anyway, I've seen a lot of tests with mixdowns where A has been mixed down "clipping" according to indicators in Cubase/Logic (but master fader set so the master doesn't clip) and B has been mixed down without clipping on any individual channels. These files sum to zero (phase reversed) when volume has been adjusted, which means that there isn't ANY difference at all between the two ways of mixing. And that's a fact. Proven theoretically and in real life.

However, you can certainly clip a signal when you export/bounce it (and that's something different), because now you're writing to a file with a certain bit depth. You should avoid anything that peaks to more than -0.1 dbfs when exporting, just to make sure you're not clipping the signal. Oh, and mp3 is another story. When you convert a file to mp3, you should avoid that the signal peaks higher than -0.4 dbfs. This is due to the mp3-codec using some bits for other "stuff" than pure audio. So if you take a "fully" peaking wav-file and convert it to mp3, then yes - it WILL clip! And it sounds awfull!!!

On another note, it might be the best thing to get use to mixing without any clipping indicators lighting up anyway (even when it doesn't affect the sound). One day, you might be mixing on a system that uses fixed point processing in the signal path, and then you suddenly have to change your bad habbits

Regards

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RiValMan

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Last edited by RivalMan on Sep-22-2006 at 15:09

-0.4 db peak sound awfull in mp3?

I don't know I was taught to master to -0.1 and never had a problem

maybe it's just that you're using a shitty codec for encoding?

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quote:

Originally posted by BOOsTER -0.4 db peak sound awfull in mp3? I don't know I was taught to master to -0.1 and never had a problem maybe it's just that you're using a shitty codec for encoding?

Not really. LAME and Fraunhofer both reveal the same results. And to be specific, it's not -0.4 dbfs, but anything higher than -0.3 dbfs. I just added an extra 0.1 to be on the safe side.

But I would be quite interested in hearing about an mp3 codec that doesn't suck (in general) and in particular an mp3 codec that doesn't clip signals higher than -0.3 dbfs. The test is easy to do:
Master something up to -0.1 dbfs and encode it to mp3. Now open the mp3 in a wave editor and see if anything has been clipped. I'm pretty sure that you'll see it has been.

I don't do mastering myself, but if I did, I wouldn't go higher than -0.4 dbfs anyway. A lot of shitty players out there distort the signal if it's too hot. And with the dynamic range available in digital today, I don't think the last 0.4 db are that important.
Except if you specifically want your track to be louder than anything else out there ;-)

Regards

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RiValMan

- Pumping the bits using:
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Thanks guys, got all feedback I needed to continue mixing the track =).

About the mixing, I was staring myself blind at -6 db befor, but it didn't work for me. My mixing improved alot when I stopped and just tried to listen to the sound instead.

However, when the song is finnished I'll try to make sure the masterchannel peaks at around -3db. I've heard this is good, because this leaves some space for the mastering engineer to work with... if you ever go that far

Cheers!

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