Automatic Zero dB gain ?

[JJGreen]

I'm setting up a template for regular tasks, and would like to know if there is a way to set up a blank project track to automatically raise (or lower) the peak levels to always be 0 dB for any stems I import.

I presently have Selig Gain, Leveler and Kilohearts Gain in my Rack Extension collection

[selig]

I'm setting up a template for regular tasks, and would like to know if there is a way to set up a blank project track to automatically raise (or lower) the peak levels to always be 0 dB for any stems I import.

I presently have Selig Gain, Leveler and Kilohearts Gain in my Rack Extension collection

No, but you can import your audio then normalize it all to 0 dBFS. BUT, if you do that you'll clip the mix like crazy when you play all the stems together.

What I do when importing audio is to normalize all audio clips THEN lower the Clip Gain by 12 dB to leave headroom for mixing.



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[JJGreen]

Thank you, very helpful and that sounds like a good tip. I can't launch Reason right now as I'm working in another DAW today, but assuming it's a simple right click menu command or something. I'll work it out.

That would also mean that when using the Leveler, I won't have to bother finding the highest peaks to read, because I'll know they are 0 db. That literally slashes this part of my processing by 50%

[JJGreen]

As a side comment, your Leveler is my most often used rack extension out of the 40 or so that I have installed.

I recommend to any forum readers who haven't used it to take a look at what it does, and think about how this can be applied to their projects.

[selig]

Thank you, very helpful and that sounds like a good tip. I can't launch Reason right now as I'm working in another DAW today, but assuming it's a simple right click menu command or something. I'll work it out.

That would also mean that when using the Leveler, I won't have to bother finding the highest peaks to read, because I'll know they are 0 db. That literally slashes this part of my processing by 50%

Yes, makes setting the Target on the Leveler super simple!

One thing you have to do is to make sure to set the SSL channel routing so that the Insert section is FIRST. Otherwise, your levels into the Leveler will be affected by the channel EQ/Dynamics (if used).



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[JJGreen]

Cheers - since we're on the topic, how does Leveler process audio below that of the low reading, does it leave it untouched? I would have though if this happened, I would notice.

Maybe I should make my question a bit more clear, because I don't have the technical understanding to express what I mean in proper terminology

Starting with a 0 dB peak, if I take the low reading of let's say -10 db and set the gain change +10 dB to compensate, what happens to parts that might be -15 dB? Is this incrementally raised by the same gain change as well?

And if so, for maintaining a dynamic range, maybe it's actually better NOT to use the lowest possible audio reading.

[selig]

Cheers - since we're on the topic, how does Leveler process audio below that of the low reading, does it leave it untouched? I would have though if this happened, I would notice.

Maybe I should make my question a bit more clear, because I don't have the technical understanding to express what I mean in proper terminology

Starting with a 0 dB peak, if I take the low reading of let's say -10 db and set the gain change +10 dB to compensate, what happens to parts that might be -15 dB? Is this incrementally raised by the same gain change as well?

And if so, for maintaining a dynamic range, maybe it's actually better NOT to use the lowest possible audio reading.

Leveler works like this (using your example):
If your peak is at 0 dBFS, then you set your Target to 0 dBFS. If you then set your Curve to 10 dB, then anything at -10 dBFS (Target minus Curve) will be raised to the same level as the Target. ALSO, everything another 10 dB below THAT will transition back to unity, leaving everything from -20 dBFS absolutely untouched.

As for dynamic range, if you set the Blend control to 100%, you get exactly what I describe above. Obviously if you set Blend to 0% you get the original dynamic range. Setting Blend to 50% leaves roughly half the dynamic range as before, which is how I most often use Leveler. In fact, in some cases I use even lower values of Blend.

So it is the Blend control (much like the Ratio control of a downwards compressor) that determines overall how much dynamic range is reduced. So it's not the Curve control that determines how much you affect the dynamic range - Curve controls how much of the dynamic range you "lift" towards the Target. Make sense?


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[afrodude]

I would setup a Template using the Gain Knobs at the Top of the SSL and set them all to something like -10 dBFS or -18 dBFS and the simply import the files select them all and hit normalize.
By using that template you can simply normalize all your stems once you have imported them using a keyboard shortcut and two clicks while making sure that you dont clip.

[selig]

I would setup a Template using the Gain Knobs at the Top of the SSL and set them all to something like -10 dBFS or -18 dBFS and the simply import the files select them all and hit normalize.
By using that template you can simply normalize all your stems once you have imported them using a keyboard shortcut and two clicks while making sure that you dont clip.

There are a few things that make this approach more work than what I suggested. First - how many channels in your template? If you need more you need to create them and then set the input gain to match the other channels. You'll also have to spend some time creating the initial template, setting input levels to the desired level (clip gain allows you to type decibel values directly). Finally, if you need to raise/lower all the levels at any point, there's no way to do this (with clip gain it's two steps - select all and type in a value.

PS, you'll also run into this issue if you create a parallel channel, having to be sure to set the input gain on any parallel channel you create to match the input gain on the original. Again, feels like selecting all clips and being able to type in any level directly is much simpler and foolproof than using Input Gain for this.



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[JJGreen]

In the application my query was regarding, it's always only three tracks. No parallel processing.

[selig]

In the application my query was regarding, it's always only three tracks. No parallel processing.

When mixing few tracks you can get away with a higher peak level on each track. Using clip gain allows you to quickly adjust all clips together so that your mix level hits where you want. You may find that with only three tracks you can set them higher than -12 dBFS (try -9 dBFS for starters).


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[Marco Raaphorst]

0 dB is tricky. MP3 compression clips at 0 dBFS. YouTube, Soundcloud, all tricky at 0 db. Use something like -1 db and you'll be safe.

[selig]

0 dB is tricky. MP3 compression clips at 0 dBFS. YouTube, Soundcloud, all tricky at 0 db. Use something like -1 db and you'll be safe.

In this case we're discussing tracks you import for mixing, not export for mastering or to publish. Since the normalization is done in the floating point world, then reduced back down again to a set level, there's no concern about data compression issues IMO.


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[Marco Raaphorst]

0 dB is tricky. MP3 compression clips at 0 dBFS. YouTube, Soundcloud, all tricky at 0 db. Use something like -1 db and you'll be safe.

In this case we're discussing tracks you import for mixing, not export for mastering or to publish. Since the normalization is done in the floating point world, then reduced back down again to a set level, there's no concern about data compression issues IMO.


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I never master, I always export from either Reason or Ableton Live directly using a fixed -1 db limit.

[selig]

0 dB is tricky. MP3 compression clips at 0 dBFS. YouTube, Soundcloud, all tricky at 0 db. Use something like -1 db and you'll be safe.

In this case we're discussing tracks you import for mixing, not export for mastering or to publish. Since the normalization is done in the floating point world, then reduced back down again to a set level, there's no concern about data compression issues IMO.


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I never master, I always export from either Reason or Ableton Live directly using a fixed -1 db limit.

Again, just to be clear, this thread is about IMPORTING audio into Reason for MIXING.

[JJGreen]

Again, just to be clear, this thread is about IMPORTING audio into Reason for MIXING.

It's funny to hear you say this, as I don't consider what I'm doing in this application to be 'mixing'. Well I guess it is, but it's incredibly rudimentary.

Track peaks at 0dB TICK
Master average to be around 0db TICK
Add intro/outro soundbytes
Export to WAV, and job in Reason is finished

To clarify (since you talked about pulling everything down to -12 as a good mixing start point with lots of wiggle room), I don't think this even applies to me. This job is just conversations between two people, so there are rarely going to be audio playing simultaneously, unless they speak over each other. And I would say that then is only going to be a conflict if both tracks, at the exact same instance, are hitting their individual 0db peaks.

Since we're here.......... As I wrote the above paragraph I jumped to a conclusion that on two tracks, even if they both peaked at the same time at +1db, the master would be +2db

But I was wrong ! According to an online calculator, it's +4db ! I guess that because 0dB is not a 'zero' value like other scientific and mathematical standards, but actually has a value.

Only confusing myself more though, when I add 0db + 0db it equals 3

http://www.sengpielaudio.com/calculator-spl.htm

[selig]

Traditionally there's tracking (recording), overdubbing, mixing, and mastering. Out of those choices, I'd say mixing is the best option for what you're doing, even if you're only mixing one track (a solo piano, mixing dialog for video, etc). Not a perfect description, to be sure.

Does 0 dB have a value? Not as written: decibels are a ratio, so 0 dB on it's own means "no change" or unity gain. If you add 0 dB to an audio signal, there will be no change.

BUT, when you specify a reference so as to create a ratio, you CAN use just "dB" to describe level, as in "add 6 dB". When specifying a level, you have to reference it to "something". In the case of digital levels, we use "FS" (full scale) as the reference point, such that if I say the peak level was -12 dBFS I've specified two points: the level is 12 decibels below full scale (clipping). One more thing: there are multiple metering options, so if I use RMS or VU I will be describing a different actual level as when I use Peak metering. So the complete description would be something like "-12 dBFS Peak" or "-20 dBFS RMS".

As for adding decibels/channels, it's impossible to say EXACTLY what the results will be in any one case, unless the two sources are 100% coherent . IF you added two of the exact same signals, say, when you create a parallel channel in Reason, the level will increase by 6 dB. This is why you should lower the level of both channels by 6 dB when creating a parallel channel (to keep your original level in place).

BUT, when you add two similar channels, or two totally different channels (aka "incoherent" or "non-coherent" sources), it can add OR subtract level because audio signals are bi-polar and have both positive and negative values. It is for this reason it is often assumed that when you add two channels together, the AVERAGE increase will be 3 dB. This is just an estimate, and assumes each channel is playing audio and is at a similar level to begin with (there are obviously a lot of variables here!). Since it's an average, it means that at SOME points you could have 6 dB increase when adding the second channel (never more than 6), but at other points you could have 3 dB, 0 dB, -4 dB, etc. so it depends on when you happen to measure the signal.

Note that when you add two channels, you get an average of 3 dB increase, but you have to add FOUR channels to get the next 3 dB increase. This continues, such that adding 8 channels gives you 9 dB on average, 16 gives you 12 dB, etc. This assumes all channels are playing audio and are at around the same level at all times. In a typical mix, not all channels are playing at the same time, and not all channels are at the same level (if you solo them one by one you'll see that). This is why I choose to leave around 12 dB of headroom on all channels, as that tends to work out nicely for most mixes I work on.

In a related subject, this is why -3 dB is often chosen as pan law value, so that when you are panned center you will be reducing both L and R by 3 dB to prevent a "bump" in the middle position. But some pan laws (the original SSL) choose 2.5 dB, some choose 4.5 dB, and some give the option to choose the one you prefer for the job at hand (or personal preference).

Also related: in digital audio, each "bit" gives you 6 dB additional dynamic range to work with, so 16 bit audio has 96 dB dynamic range (6 x 16 = 96, and 24 bit gives you 144 dB. Again, this relates to how doubling the voltage (adding two identical signals) adds 6 dB gain, making "6 dB" a "magic number" of sorts in the audio world. This is also why level scales are often (but not always) a multiple of 6 or 3 dB, in EQ gain boost/cut, or fader values as marked on the fader (as two examples).


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