Mixing question........levels

[samsome]

so i bring all faders down.......before i start mixing

so as i raise sounds up its possible i hit 0 before even raising all tracks up.....so each time i hit 0 do i just lower them all the tracks together at same level? and then carry on from there again?

is that how you do it?

[Loque]

I take the advice from Selig, and thats keep all sounds around -10db when leaving the rack and than go into mixing (hopefully i said it wrong or Selig will salp me!).

Than just do your mix. You have enough headroom if something will peak and the final compressor/limiter will catch it.

[samsome]

I take the advice from Selig, and thats keep all sounds around -10db when leaving the rack and than go into mixing (hopefully i said it wrong or Selig will salp me!).

Than just do your mix. You have enough headroom if something will peak and the final compressor/limiter will catch it.

hm sounds like a good idea, thanks!

btw did you mean rms -10db? or peak?

[guitfnky]

I don’t worry about levels when they’re in the box, unless something is distorting (this is almost always due to pushing an insert effect with a hotter signal than it should have—something that fader moves won’t help with). as long as everything sounds good to your ears, and your master output isn’t clipping, you’re fine.

here’s a good explanation of why it’s not something to be super concerned with:

[sdst]

[Billy+]

so i bring all faders down.......before i start mixing

so as i raise sounds up its possible i hit 0 before even raising all tracks up.....so each time i hit 0 do i just lower them all the tracks together at same level? and then carry on from there again?

is that how you do it?

Give these a watch
https://youtube.com/playlist?list=PLljy ... JNhp3tAtPZ

[Tiny Montgomery]

I don’t worry about levels when they’re in the box, unless something is distorting (this is almost always due to pushing an insert effect with a hotter signal than it should have—something that fader moves won’t help with). as long as everything sounds good to your ears, and your master output isn’t clipping, you’re fine.

here’s a good explanation of why it’s not something to be super concerned with:

This is excellent.

[mcatalao]

No. The way i do it is, start from unity (faders at 0 dbfs) and make the peak level of each channel be around -12 to -18 dbfs (clip).

For that you can use a Selig gain inside your input or use the gain knob on the top of the mixer. Always control the peak level with the peak monitor (I always have it at the VU + Peak mode, but don't let the VU meters at the channel level fool you). So with the mixer Gain knob i take care of absolute peak levels, then with the faders I take care of apparent levels and relations. When you add stuf to your inserts, be sure to check levels if the devices don't adjust automatically.

[selig]

Gain Staging is an analog concept, no longer relevant in the digital (floating point) world.

In making the change from analog to digital tape and then to digital fixed to digital floating point recording in DAWs, I've had to adapt my workflow over the years.

One major change has been how to deal with levels in a DAW, now that we have such incredible dynamic range available compared to analog and even 16 bit digital recording. So instead of traditional gain staging I have adopted using a consistent peak level for all audio signals. This means that the source (software synth or external audio) hits a certain level which is maintained "consistently" up to the channel fader.

The "myth" is that "gain staging", which today actually just means keeping control/awareness of levels, makes your mix "sound better". The truth is it simply makes mixing easier, which actually CAN make your mix sound better. Compare the scenario you describe, adding tracks/pulling down master fader, repeat, to an alternative approach which is to START with the levels reduced ahead of time. In the second scenario you are simply putting the faders/levels where they sound balanced, knowing the mix isn't clipping. No looking over your shoulder, no going back and forth each time you add a new track to the mix - just "mixing".

Why does this happen in the first place? Think of it this way: if you start with all your individual audio tracks at the highest level possible, hitting just below clipping, you can only add (sum) a few together before you clip the mix. Imagine mixing a drink using full glasses of each ingredient, and pouring them all into a similarly sized glass - you quickly overflow the glass. Same with audio signals, often mixing 30-40 or more tracks together. The difference is that not all tracks play at exactly the same time, and not all tracks will be "mixed" at full level. So in some cases you CAN get a handful of tracks to combine nicely and not clip, but probably not more than that before you exceed the headroom of the mix bus.

Luckily in a floating point system you have a huge amount of headroom, so you CAN just keep pulling the master fader down as you add more and more tracks to your mix. BUT, you can also just build in the headroom from the start. Which workflow you choose is entirely up to you IMO. But once I started using a consistent peak level for all my audio tracks, I found additional advantages…

Advantages to mixing with consistent peak level tracks:
•Leaves headroom for the mix bus, the initial advantage being discussed
•Leaves headroom when recording raw audio
•Allows A/B comparison of added processing, since the levels coming in to every device match the levels going out
•Allows bypassing/deleting any added processing without changing overall track levels, thus preserving your basic mix balances
•Removes any question of "what level is this signal" at every point in the signal path, since you already know the highest peak level
•Makes setting up any non-linear processor (dynamics, saturation, distortion) easier because you already know how hot the input signal is

All in all this can contribute to a faster/simpler workflow, and less time "fixing" equals more time mixing! My "system" that I've evolved to involves setting sources to peak around -12 dBFS. I typically mix 10-40 tracks or so, no more, so this level works great. This gives a VU on average of -24 to -18 dBu, which is what most analog simulation plugins are expecting. Remember that for most audio interfaces, 0 dB VU from the analog world equals -18 dBFS in the digital world, set using a sine wave test tone typically @ 1kHz.

I made this video to show the process in action.

Note that I don't need to adjust the Input Gain on the SSL mixer because I already set levels from the very first source - one of the keys to making this approach work across the board. Also note that once you combine signals, such as a drum bus, the levels will naturally be higher - this is normal, the process I'm using only applies to individual tracks, not stems. So I end up with tracks peaking around -12 dBFS, stems (sub-mixes) peaking around -9 dB or slightly higher, and aim for a mix peaking between -6 and - 3 dBFS and a master peaking at -0.5 dBFS. Make sense?

[illlumen]

Note that I don't need to adjust the Input Gain on the SSL mixer because I already set levels from the very first source - one of the keys to making this approach work across the board. Also note that once you combine signals, such as a drum bus, the levels will naturally be higher - this is normal, the process I'm using only applies to individual tracks, not stems. So I end up with tracks peaking around -12 dBFS, stems (sub-mixes) peaking around -9 dB or slightly higher, and aim for a mix peaking between -6 and - 3 dBFS and a master peaking at -0.5 dBFS. Make sense?

Very well said. You explained it really good, I couldn't have done it better!
Sometimes I am using a VU meter plugin to check if my levels, too.
I have set my speaker gain in a fixed way so my ears are finally able to hear the right levels.
Having enough headroom (and a correct workflow) is something I had to learn the hard way.
Quite happy enjoying the results! It makes life much easier for me.

[selig]

]
Very well said. You explained it really good, I couldn't have done it better!
Sometimes I am using a VU meter plugin to check if my levels, too.
I have set my speaker gain in a fixed way so my ears are finally able to hear the right levels.
Having enough headroom (and a correct workflow) is something I had to learn the hard way.
Quite happy enjoying the results! It makes life much easier for me.

I would not suggest using a VU meter to check levels when headroom is a concern. This is because VU metering shows “average” levels, and the actual peak level can be as much as 20 dB (or more) above the VU level in some cases. Meaning, even if you use -18 dBFS as your reference level but measure in VU, you can still very easily be clipping the output (sometimes with only a single channel playing).

A real world example would be trying to get out of a speeding ticket by arguing your AVERAGE speed was well under the speed limit...

[guitfnky]

A real world example would be trying to get out of a speeding ticket by arguing your AVERAGE speed was well under the speed limit...

/digression
I’ve always imagined getting pulled over for speeding on an otherwise empty stretch of highway and when asked for an explanation, saying “public safety, sir/ma’am—just trying to keep up with traffic”, and gesturing vaguely at the road ahead.

I’m sure it would go over as well as I imagine.

[motuscott]

I’ve always imagined getting pulled over for speeding on an otherwise empty stretch of highway and when asked for an explanation, saying “public safety, sir/ma’am—just trying to keep up with traffic”, and gesturing vaguely at the road ahead.
I’m sure it would go over as well as I imagine.

Wouldn't try this in the US of A these days. Police force a little twitchy.

[DJMaytag]

The minute you start using plugins that are analog modeled or do anything non-linear (like saturation type plugins), you absolutely do have to worry about the signal levels going into those plugins, which means paying attention to the levels of other things in the chain. Emulations of hardware (like the 1176, LA-2A, LA-3A, etc) mimic the distortion that occurs at certain levels, so you need to reference similar levels in the box, or else you will get VERY audible aliasing distortion (unless you run your audio at 96kHz or higher, and/or have plugins that oversample/allow oversampling).

[DJMaytag]

The "myth" is that "gain staging", which today actually just means keeping control/awareness of levels, makes your mix "sound better". The truth is it simply makes mixing easier, which actually CAN make your mix sound better.

I had a lot of hesitation jumping fully from Cubase to Reason 9.5, mainly due to how I would frequently enter very exact numbers into the mixer settings in Cubase. I was worried about not having that same degree of accuracy in Reason, but I've found that NOT having the fine degree of control has made my mixes better. Similarly, following a system where I'm setting my levels lower like you've mentioned has made the mixes easier & better too.

Note that I don't need to adjust the Input Gain on the SSL mixer because I already set levels from the very first source - one of the keys to making this approach work across the board. Also note that once you combine signals, such as a drum bus, the levels will naturally be higher - this is normal, the process I'm using only applies to individual tracks, not stems. So I end up with tracks peaking around -12 dBFS, stems (sub-mixes) peaking around -9 dB or slightly higher, and aim for a mix peaking between -6 and - 3 dBFS and a master peaking at -0.5 dBFS. Make sense?

Sometime I do adjust the input gain, but yeah, starting at the source is a really good idea. I usually start all my patches from scratch, so I'm not having to constantly fight the "presets loudness war" that is a PITA on some instruments. I pretty much always bring Europa down to 0db, instead of its default +8.8db setting, for example. ABL3 is another one (the VST) that's got a hot output, so I'm often putting its volume knob at 40-50%, instead of its default 80% volume setting.

I usually strap a LUFS meter to my main insert, which tells me a lot more about my signal that dBFS numbers. I have my monitor setup (w/ stepped volume controller) calibrated to -18 LUFS which yields a volume of 83dB-C. That's a pretty solid listening volume that can be tolerated for a decent amount of time without fatigue, and is a volume where your ear hears everything as being relatively flat. This calibration is arguably more critical to getting good mixes than is gain staging.

[illlumen]

A real world example would be trying to get out of a speeding ticket by arguing your AVERAGE speed was well under the speed limit...

I understand. The VU meter isn't the only one I am keeping an eye on. I forgot to mention that I always have an eye on the peak meter, too. I have set my listening levels to a fixed setting, so my ears are tuned well by now. Hearing inter sample peaks is something different, though. Really funny example of yours! I like your way of explaining - and your humor, too.

[Kalm]

Yea.... I stopped gain staging in Reason. I just focus on making the sounds sound gooder. I'll many times crash my peak into the 0 dB realm (mostly drums) but it helps to hear what the clipping/limiting stage may sound like. I just simply make sure my channels don't clip. Every once in a while I'll gain my kick down to my norminal level -12 dBFS and adjust all sounds accordingly when I'm ready to export/mix in S1. Other than that I let my music breathe when producing, slap a limiter on it , export and call it a day until the mix session

[selig]

I just simply make sure my channels don't clip.

Channels CAN'T clip - only the final output to your D/A (or file export) can clip.

[Billy+]

If you open the file in an app capable of displaying the wave form and the peeks look right at the top rather than a bit of space left then you probably caused the audio to be clipped and if you can hear it on the speakers / headphones and you will then you have.

Also if you haven't and you decided to do any kind of encoding like wav to MP3 and your just shy of clipping chances are the encoding process will cause the new file to clip...

[sdst]

I saw that video that cliping everywhere doesn't matter, when i do that my mix sounds like crap

I stick with the old way, -15 pk in track then there I sound good

[selig]

I saw that video that cliping everywhere doesn't matter, when i do that my mix sounds like crap

I stick with the old way, -15 pk in track then there I sound good

You can't clip "everywhere" - in fact there is only one place you CAN clip - the final output.
That's why clip indicators appear ONLY on the final output meters, not on channel meters!

[sdst]

You can't clip "everywhere" - in fact there is only one place you CAN clip - the final output.
That's why clip indicators appear ONLY on the final output meters, not on channel meters!

I understand that the red in the channels is a decoration, that they should remove that red then

my mix doesn't sound good if all channels are in that red, I stay the old way like i'm in the 80s

[mcatalao]

A real world example would be trying to get out of a speeding ticket by arguing your AVERAGE speed was well under the speed limit...

[integerpoet]

…clip indicators appear ONLY on the final output meters, not on channel meters!

I understand that the red in the channels is a decoration, that they should remove that red then

It does seem from a human interface design perspective that channel meters should be a gradient from green at the bottom toward yellow at the top. I'm sure RS is listening carefully to this thread and will rectify this in R12.

[selig]

You can't clip "everywhere" - in fact there is only one place you CAN clip - the final output.
That's why clip indicators appear ONLY on the final output meters, not on channel meters!

I understand that the red in the channels is a decoration, that they should remove that red then

my mix doesn't sound good if all channels are in that red, I stay the old way like i'm in the 80s

It's not because channels meters are in the red that your mix doesn't sound bad - for one, you can turn that feature off so the VU meters show green all the way up - it's TOTALLY arbitrary! For another, you can lower the master fader and get totally rid of any clipping, so again it's not the channel meter levels/colors causing a mix to sound bad.

Remember that the VU meters (that show red) are AVERAGE reading. Peaks can be WAY higher than the VU meters show because the peak level is always higher than the VU level. You can potentially have a single channel TOTALLY CLIPPING the output with the meters showing GREEN. So "GREEN" means as little as "RED" with the VU meters, sorry to say.
And 100% yes, they should get rid of the red option for the channel meters. Again, you can so easily turn this off it it distracts you…