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The statement below was found in the notes for a beginning mixing class. Agree or disagree and why?
Recording/mixing at well below max level (-10 dBFS to -20 dBFS) allows for louder and cleaner masters. After all, the end product is what matters most. The old axiom of “use up all the bits” no longer applies in the age of 24-bit recordings and gear with lower noise floors.
Jim Fogle - 2025 BiaB (Build 1128) RB (Build 5) - Ultra+ PAK DAWs: Cakewalk Sonar - Standalone: Zoom MRS-8 Laptop: i3 Win 10, 8GB ram 500GB HDD Desktop: i7 Win 11, 12GB ram 256GB SSD, 4 TB HDD Music at: https://fogle622.wix.com/fogle622-audio-home
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The statement below was found in the notes for a beginning mixing class. Agree or disagree and why?
Recording/mixing at well below max level (-10 dBFS to -20 dBFS) allows for louder and cleaner masters. After all, the end product is what matters most. The old axiom of “use up all the bits” no longer applies in the age of 24-bit recordings and gear with lower noise floors. It's a true statement but is more applicable to a beginner mixer. A good, clean recording at a higher level will require less, if any effects such as compression, eq and limiting to bring the dynamics up to level. It's always better to get the recording right than to fix it during post. With beginning mixer, it's also better to be safe than sorry so there is something to work with if the artist is gone and the recording is crud.
BIAB 2025:RB 2025, Latest builds: Dell Optiplex 7040 Desktop; Windows-10-64 bit, Intel Core i7-6700 3.4GHz CPU and 16 GB Ram Memory.
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Disagree:
You want to have the levels set just far enough down so that you don't exceed 0dB and venture into clipping. -6dB for peaks is fine. But you could push it up further if you needed to. I tend to work around -3dB on peaks.
If you record too low, yes, you can turn it up, but, when you do, you also turn up the noise and hiss that is in the track. As you turn it up, you raise the noise floor as well. -20dB is way down. That level is lower than where I run my vocal doubling secondary tracks and you can't hear them unless you solo the vocal buss. They are not recorded at that level..... they are recorded up just under 0dB, but I pull the faders back for the final mix.
You want to have the strongest signal level going in that you can possibly get without clipping. That keeps the source material well above the noise floor.
I prefer to have nice strong levels in the tracks and then pull them down for the final mix if that's needed, rather than going the other way.
Just my 2 cents
You can find my music at: www.herbhartley.comAdd nothing that adds nothing to the music. You can make excuses or you can make progress but not both. The magic you are looking for is in the work you are avoiding.
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I agree Herb. I try for around -6 db too.
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Well....
We all have our particular recording protocols that work for us.
For me...I've always recorded individual tracks to about -12db...give or take a couple....and I've never had any level issues at any stage.
Therefore, I see no benefit in recording at a higher input level.
Back in 2010 I'd sent (3) of my tunes to be mastered at Buffalo Sound (now called the The Ranch I think).
He said to send all raw tracks at -12db.
(He also added lead axe and B3 tracks which was very cool)
Most of us are in the digital realm so noise should not be a big issue as far as recording levels.
So...that recommended level (-12db) has more to do with the head room comfort necessary with summing of all tracks along with FX/signal processing at the stereo mix.
If you're using tape, different story, then as close to 0db is best but you can even push that level in the tape realm without issues.
If the end result has been achieved regardless of the path chosen all is well.
That's my take on it.
Carry on....
Last edited by chulaivet1966; 09/20/15 04:40 PM.
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IF you are using plugins that emulate vintage hardware and you are recording/mixing in 24bit, then the "sweet spot" is said to be -18dbfs and it's suggested to use the trim on your DAW (or a trim plugin first in your fx chain) since the signal is prefader going into the plugins. This is from Graham at the Recording Revolution: http://therecordingrevolution.com/2013/11/25/do-you-know-how-to-read-your-meters/And I've read long discussions about that from some other places too. But personally, I just keep the peaks somewhere between -12 and -6. I use Realtracks and rarely record anything other than vocals. And i found the above advice (-18dbfs)to not work well for me mixing with the FX I use. YMMV
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Jim,
Thanks for starting this thread. I'm finding it very interesting reading.
Noel
MY SONGS...Audiophile BIAB 2025
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IF you are using plugins that emulate vintage hardware and you are recording/mixing in 24bit, then the "sweet spot" is said to be -18dbfs and it's suggested to use the trim on your DAW (or a trim plugin first in your fx chain) since the signal is prefader going into the plugins. This is from Graham at the Recording Revolution: http://therecordingrevolution.com/2013/11/25/do-you-know-how-to-read-your-meters/And I've read long discussions about that from some other places too. But personally, I just keep the peaks somewhere between -12 and -6. I use Realtracks and rarely record anything other than vocals. And i found the above advice (-18dbfs)to not work well for me mixing with the FX I use. YMMV Hi Josie, That's interesting that you mention -18 dBFS. I find that for my individual instrument tracks (wav), I usually have them set so that the meters, on average, oscillate around the -18 dB mark in Reaper. For my vocals, they're usually a little higher... I estimate somewhere around -12 dB. For the Master Volume, though, I set it so that it peaks a little under 0 dB. I arrived at these values purely by trial and error and I've stayed with them because they seem to work for me. (This is the main reason that I run Reaper with a version 3 skin... I like the extra height of the meters that v3 has.) Since I have Ozone as an FX in the Master fader, I find that I need to adjust the Master fader when I play around with Ozone's presets. Regards, Noel
MY SONGS...Audiophile BIAB 2025
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Just for the record. I never... well let's just say, rarely.... I rarely use numbers to determine my levels. I record the track and look at the wave forms. This is my normal looking track view. The waveforms are 75% or more of the track. If they are more like straight lines, they were too weak and get recorded again after I adjust the input level up. It's rare that I overload anything I record because I have the max input levels set in my interface's software control panel front end.  DUST ON THE FLOOR This is a screen shot closeup of the vocal from another studio. Notice the "overs" starting around the center and going to the right (forward) from that point. The interesting thing is that listening to this, the over's don't seem to wreak to much havoc in the tune. I try to avoid this kind of thing in my studio, but there's very little control over stuff that comes in the door.  IN A WORLD WITHOUT YOU I included the links to the songs I referenced so you can hear the results. The way I determine my levels is by looking at the wave forms and also by observing the meters as I'm in playback mode and record mode. I adjust levels to stay out of the red. An occasional red blink is OK but I do not want to have constant red showing up in either mode. I look at the outputs and then after a render or export, I look carefully at the final wave to see how many peaks are touching 0dB both in the original wave and in the converted MP3 file. Just for grins, I set my levels up a bit and exported this. This is what I don't want to see. 
You can find my music at: www.herbhartley.comAdd nothing that adds nothing to the music. You can make excuses or you can make progress but not both. The magic you are looking for is in the work you are avoiding.
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Does anyone want to learn some of the terminology involved and how that matters for setting input levels? If so, this is what I picked up in EE-638 from Purdue University many moons ago.
The quantization signal to noise ratio is 48 dB better in a 24 bit recording compared to a 16 bit recording. The rule of thumb is that the QNSR improves 6 dB for every bit of the A/D conversion at hand. To hear how this works in the reverse, take a nice clean 24 bit signal and then do bit reduction to 16 bit (most won't notice), then bit reduce to 12 bit (you will probably hear this) and then 8 bit, and you'll definitely harken back to the old Castle Wolfenstein game days with it's crudely rendered words and sounds.
Quantization noise is what happens with the signal that is effectively zero and the A/D has to decide if the signal is actually all zeros, or all zeros except for a 1 in the LSB.
2^16, or 65,536 'shelves' to stick the signal on with 16 bit recording, and 2^24 or 16,777,216 potential shelves to stick the signal on with 24 bit recording.
The quantization of the signal for a 24 bit at such a finer resolution is what accounts for the 48 dB better QSNR.
Translation: There is no harm in recording near full-scale use at 24 bit resolution, but there's also WAY WAY WAY more wiggle room (maybe another WAY in there) with 24 bit vs. 16 bit. I state that there's no harm because you can scale down the full scale tracks after you've made the recording, but the chance to be introducing quantization noise at let's say -18 dBfs, is for all intents and purposes non-existent.
Last edited by rockstar_not; 09/21/15 04:27 PM.
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I'll pass on the big fancy words. As long as I can get the results I'm looking for..... "a couple dB lower or higher" works for me". 
You can find my music at: www.herbhartley.comAdd nothing that adds nothing to the music. You can make excuses or you can make progress but not both. The magic you are looking for is in the work you are avoiding.
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