The BEST WAY to drop VOL on all tracks

Hi all.

I read another thread somewhat considering this issue and I wanted to start a whole new thread since this is bugging me in Cubase the MOST; That is:

How to adjust the gain of all the audio tracks simultaneously regardless if there is automation present or not??

You tell me that. I have found no way. I guess you can use the Automation Panel but I have been too lazy to even learn to use it properly. So if you can use that, then it’s okey but someone said on the other thread that you use AP to “all tracks with automation”. That’s just it. I don’t want to check every single track whether it has volume automation or not - I want to be able to set the volume on ALL AUDIO TRACKS IN ONE GO. Whether there is vol automation present or not.

I find this very disturbing thing in Cubase. I have started now the new projects by first lowering the volume of each track to -12dB before I do anything else. But STILL I seem to get to the point where the master volume tops 0dB.

Could the input gain knob of the master track be used to adjust that? I would guess so. Or does it STILL mean that at some point the sum of all the tracks would still produce distortion? I mean if any single track is not producing distortion but the sum of all the tracks do, then does the master track input gain provide the solution? Or will the sound get distorted at some point of the prosessing line? Repeating myself, sorry…

Anyway, I have mentioned this before. There once was a tiny audio software named SAWPLUS32 where you had the button named “OFFSET” beside each and every audio track. When that OFFSET button was down (activated) one could easily adjust the overall volume of the track regardless of if there was any automation present or not. Now that is a feature that I would love. And yes, there is probably a way of doing it with Automation Panel but again - I’m too lazy to go and learn it. I just hate reading manuals. So if someone could write the instructions here in the forum I would read it with pleasure, however :wink:

-Tommi

I got it (I think) !!!

I did the following to adjust the volume on ALL tracks regardless if there are automation or not - this works in ANY case:

  • First link all the audio tracks
  • From the Automation Panel (AP) click “enable write on all tracks”
  • From AP enable the two choices on the left: TO START and TO END
  • From the AP enable TRIM function
  • From the mixer change the volume of one of the linked tracks

Voilá. It changes the volume on all tracks regardless of any automation present or not.

What do you think? Of God… Quite a hassle with this one but nonetheless it works.

-Tommi

Hi all !!

I got a simple and proven solution: Just adjust the master output track INPUT GAIN.

I made an experiment; Put the same audio track as 12 instances so they were all playing together. The output of 12 tracks playing the same audio material will of course 12-fold the output gain. OK, it was distorted as hell. And that’s ok.

Then I adjusted the Input Gain value to about -25dB and tried again. No distortion at all. See the pictures below. The second picture presents the downmixed audio file of all the 12 tracks playing together with the master track Input Gain value adjusted to -25dB. As you can see, the signal is not distorted at all. And it isn’t when I listen to it.


OK, I tried the same with Input Gain at 0dB and adjusted the level of the master track volume and it isn’t distorted anyway even if the 12 tracks are all “playing” together. So I guess that the master track volume adjusting is enough. Well, simple as that. Experimented. Myth busted.

Thanks Tommi - this is very useful! I’d sort-of forgotten about the input gain knob on the master fader (!), but that is the perfect way to compensate for gradually building-up of levels.

My problem with the grouping of all tracks was of course that if you’ve got any audio sub-groups going on, it lowers the levels twice (once for the individual channels, and then again for the sub-group), and with some fairly complicated routings going on using multiple sub-groups, unpicking all that so I knew just which channels to lower was a bit of a nightmare.

But your solution should solve all that. I suppose Cubase has theoretically got incredibly high internal headroom due to the 32bit floating point maths used, so it’s not a problem at all to lower it at the point of final summing, and this still leaves the full movement of the master fader available - I was wary of running the track with the master fader way down low all the way through.

Excellent! :smiley:

  • 8-bit integer: 48 dB
  • 16-bit integer: 96 dB
  • 24-bit integer: 145 dB
  • 32-bit floating point: near-infinite dB

Cheers,

Chris

I don’t know why people are affraid of using the Master Bus when there is a reason for it to be there in the first place (to lower the volume of all tracks at once before output). In Sonar, I used to use a SubBus for this purpose because it doesn’t have Post fader inserts like Cubase. And I did the same in Cubase until recently, when I learned about Post fader slots (great feature btw!). Now I don’t need a Sub Bus anymore :wink:

As Chris points out, you have near infinite resolution inside of Cubase. The only reason why we hear distortion on output is because of the resolution of our audio interfaces (24 bit integer). That’s why you have to lower the Master Bus in order to get clean playback. But inside of Cubase, there is no distortion even when you see the Master Bus going well into the red (unless you push things to unrealistic amounts over 0dB FS). Now, I’m not saying this is an excuse not to care about gain staging because it is still good practice to keep things in check. But at the same time, we worry too much about losing resolution, like in the days of analog, when those concerns do not apply to the digital world. Another one is recording everything as hot as possible, but that’s another topic.

BTW, using the Input Gain in the Master Bus is a good tip. Just keep in mind that changing the Input Gain will alter the way your inserts behave (except for Post fader inserts), so double check them afterwards to make sure they react the way you want them to (especially dynamic processors).

Take care!

Just one thing to check - I am assuming the input gain knob is after the Master summing engine but before all pre-fader Master plug-ins, so there isn’t a danger of overloading the input to any of them? Many plugins don’t use 32 bit float internal processing, so overload distortion is a very real possibility inside plugins if the level coming into them is too hot.

+1

No. You have nearly infinte dynamic range (more than 750 dB above 0dBfs, and if I’m not mistaken the same below 24 bit’s range of -144dBfs … totalling more than 1500dB). But that’s not resolution. Resolution is still dictated by 24-bit mantissa of 32-bit floating-point number. The whole beauty of floating-point audio is that you can change the level (move your fader, compress, etc) while keeping the original audio resolution.

+1 again. Only time to worry about the signal level inside Cubase’s signal path, is when there’s a plug-in which changes behaviour according to signal level. This includes:

  1. dynamic processors
  2. “vintage gear” emulations
  3. some ill-behaving plug-ins

Now when it comes to use master bus’ gain instead of fader, another case where it may be convinient to use gain if you want to do fade in/outs or other automation there.

Summary: DON’T BE AFRAID to use master bus fader to set your final levels.

Yu don’t have to assume anything. If you are not sure, just try it: put any plug-in with level meter to that 1st insert slot and chek if turning the gain-knob has any effect.

Would have been faster to test it, than write your comment :sunglasses: (sorry for rant, couldn’t resist)

@mindastray:

(googletranslate)
“viimeinen rukous” = the last prayer…?

:unamused:
I have to be worried…
:laughing:
:mrgreen:

Wow! For once the Foogle Transscribble has succeeded to produce something correct.

Where talking semantics here, but to my knowledge single and double precision math also equates to resolution. For example, the resolution of Cubase’s audio engine is 32 bit floating point. The same can be said of plugins, DSP cards, Operating Systems, CPUs, DAC, ADC, etc. Basically, when talking about bits, we’re talking about resolution.

From the Steinberg website http://www.steinberg.net/en/products/cubase/cubase6_details.html:

Next-generation audio engine
The award-winning Cubase audio engine delivers a crystal-clear 32-bit floating-point resolution and 192 kHz sample rate, with a pristine sound quality that is the hallmark of the Cubase 6 music production experience. The next-generation audio engine includes true surround capability throughout, with each track and channel offering up to six discreet channels, ready for 5.1 surround and powered by Virtual Studio Technology that opens up a myriad of VST instruments at your fingertips.

Of course we are. I find the word resolution to mean accuracy of representation. In that case 32-bit floating-point audio has 24-bit resolution. The problem here is, that in floating-point world all the bits are not equal. You have mantissa (accuracy/resolution) and exponent (range).

With 32-bit floating-point audio I can not have a sample valued 0.000000001 dB below 0dBfs, like I could if I had 32-bit fixed-point audio. That’s why I can’t say I have 32-bit resolution in 32-bit floating-point engine.

OK, I see your point. But the actual resolution of 32 bit floating-point is 25-bit (including the implicit bit) or -156dB FS. Perhaps “precision” is a better word to use when referring to floating point math. I’ll take that into account :slight_smile:

Take care!

In The Name Of The Great Pumpkin! I forgot that crucial invisible bit! Unforgivable!

There’s always one :laughing: :laughing:

Ha! You guys should sit down for a beer sometime and, yes, I would watch the video. :laughing:

No worries, man. He is a sneaky one :stuck_out_tongue:

Always a pleasure to exchange words with you, Jarno.

Take care!

The channel input gain will alter the behavior of all inserts pre- or post fader.

And (as said before) when you use non floating point plugins.

I usually mix from the ground up and, with pop music, usually hover right around 0 in the loudest part so it’s not a problem. But I often add a little 12k air and L2 compression on the master bus.

If the meters are peaking above 0, I can just turn the master fader down and use the Post Inserts for my EQ & L2?