Old 12-20-2018, 07:35 AM   #1
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Default When does -18 DB matter

How important is it to keep your faders at -18 DB?

Much of this has to do with limits of headroom old analog gear and how the mixbus interacted but often people cite that many plugins are set up for -18 and that their response will be different if you don't provide it at that level.

How true is this and how are you supposed to know if a plug-in needs -18 because the bus certainly doesn't because Reaper doesn't care as it has more hHeadroom that anyone could possibly use?
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Old 12-20-2018, 07:50 AM   #2
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How important is it to keep your faders at -18 DB?
It really only matters for the incoming signal (kind of but not really) or if you want to match the expectation of a preset in a non-linear plugin, and it doesn't really matter then sorta. The original reference to it is because of this...

In the analog world, we obviously can exceed "zero" but in digital, zero is the impassable ceiling. Knowing this, manufacturers need to account for it because if they made analog zero = digital zero, as soon as your analog signal exceeded analog zero, it would be clipping on the digital side. So, they "push down" where analog zero is on the digital scale so that from there to digital zero can cover all that >zero analog signal. With 24 bit converters many manufacturers happen to land "around" -18 dBFS. If you ran a 1k HZ sine wave into your interface line in at unity, it should show up in the DAW around -18 (or the level your sound card's manual says it uses, if it lists it).

This means that if you record and set your analog gear as you always would, there is a high chance it will show up in that -18 dBFS'ish RMS range in the digitally recorded file all by itself, with no need to adjust for that after the fact in the DAW. Once in the DAW, as I stated earlier, if you wanted to know the intention of some amp SIM preset that say's use -18 dB, it's only so that all this matches up and the preset they designed will react similarly to you as when they created it (because the sound of non-linear plugins is based how hard you hit them) - but at the end of the day, treat it like a real amp, turn the knobs until it sounds good.

One could make the point that by "gain staging" all your tracks to ~-18 dbFS RMS you have plenty of room to mix and do things that add gain as you go along with the summing of all the tracks but that's an organizational/workflow thing, not really an audio quality thing.
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Old 12-20-2018, 07:54 AM   #3
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You'll have to check the manual for each plugin, if you're worrying about how emulations are calibrated. Many have variable calibration options anyway, or at the very least an input trim knob. There is no standard for analogue modelling plugins in terms of calibration.

Personally, I don't worry about it at all. They are effects, not actual pieces of gear, so I just twist knobs till it sounds good.
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Old 12-20-2018, 08:00 AM   #4
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You can learn some here:
https://web.archive.org/web/20160102...es/digital.htm
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Old 12-20-2018, 08:28 AM   #5
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Originally Posted by Judders View Post
You'll have to check the manual for each plugin, if you're worrying about how emulations are calibrated. Many have variable calibration options anyway, or at the very least an input trim knob. There is no standard for analogue modelling plugins in terms of calibration.

Personally, I don't worry about it at all. They are effects, not actual pieces of gear, so I just twist knobs till it sounds good.
I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
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Old 12-20-2018, 08:32 AM   #6
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I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
I've seen mention of several but that's it. If it is mentioned, it's often probably for the same reasons I'm discussing - so that there is some common ground between what they heard when designing a preset and what you hear when you try it out. When I buy a real amp, sometimes they will come with these little button position charts with settings and names by them, I'd usually fark with them for ten minutes, then throw it away and turn knobs till it does what I want to hear.

Otherwise, it's like an amp, the strength of the signal hitting the amp can vary wildly based on guitar, passive, active, single, double, pedals that boost, pedals that are a boost - anywhere from say ~50mv to 1,2,3,5 volts (I built and OD that will push out almost 7 clean volts if I ask it to). What do you do then? You plug in your guitar, you turn the knobs until it sounds right.

If we always followed what the recommendation was though, guitar distortion would have never been born.
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Old 12-20-2018, 08:54 AM   #7
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Faders aren't supposed to be at -18. The levels are.

It matters for analog-modeled plugins that use that as a reference level and analog gear that uses that as a reference level, typically things that emulate tubes, transformers, or discrete circuitry....they all behave in ways that changed based on your signal's relationship to a reference level. Being wildly away from it makes them perform wildly different from the designers' intentions.

Pretty much everything else modern uses floating point processing that doesn't care about reference levels.

But, as another consequence of floating point processing, it will never harm your audio to use it. So, when in doubt, you might as well have RMS/VU meters hovering around that level on average and let the peaks go wherever they go (as long as you're not unintentionally clipping a plugin or something).
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Old 12-20-2018, 09:34 AM   #8
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I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
Here's one off the top of my head (page 8): https://www.lsraudio.com/demos/VLB525_User_Manual.pdf

There is often a little screw for calibration, or a hidden menu. If the plugin has VU meters then you can simply shoot for an average of 0 on that.

For pure digital plugins it doesn't matter, I don't know of any that aren't floating point now so headroom is of no concern.
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Old 12-20-2018, 09:40 AM   #9
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Oh, and if I'm mixing, I don't care what the levels are on individual channels. All that matters is the level hitting the master. I have Klanghelm VUMT on my monitor fx, when I bring in all the audio, I select all items and bring them down until VUMT doesn't overshoot 0 too much on the loudest parts, when calibrated to 0 dBVU = -18 dBFS, dealing with individual item gain only if relative levels are WAY off.

Obviously in a mix only the total level of all tracks matters, not what individual tracks are at.
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Old 12-20-2018, 09:48 AM   #10
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When you record an analog input with level set at around the -18db range, do you then go in and boost the volume on the recorded track by normalizing or adjusting the volume control, or do you just leave it as is?
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Old 12-20-2018, 09:56 AM   #11
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When you record an analog input with level set at around the -18db range, do you then go in and boost the volume on the recorded track by normalizing or adjusting the volume control, or do you just leave it as is?
If I want to change it, I may do it with the trim knob on the items because that will fall before any plugins. Let's say I have an -18 dbFS RMS track I just recorded, I add a compressor. Chances are if the signal is a little low for the compressor (mostly opinion), I'd bump up the trim gain to send a little more signal into the compressor just to get the level "reasonable" (opinion) - granted that isn't much different from just adjusting the threshold but for me it's helpful if the waveform reflects that change which it will if I use the trim knob. Simply because I don't magnify my waveforms, I always know how much relative output they have at the item level <-- All this usually happens very early on FWIW. Beyond all that, raise it where you need to, if the track is perfect but a little low, raise the volume slider instead.

^That difference is really more about whether you want to adjust "post recording level" or just the "relative mix level" compared to all the other tracks so short answer is whether you want to adjust before or after plugins LOL.

Tbh, when I have a handful of tracks -18 dbFS RMS might be a little low once everything sums up. Meaning that once I've processed and mixed everything, I may still only be peaking around -6 dbFS at the master. I usually make up for that on the master FX chain. IOW, I have a limiter at the end of the chain with an input volume, if I'm coming up short, I just raise that to get the peaks up closer to zero.... because I have the limiter set to not even react until it's getting into territory where peaks could go over zero; and in that case, I'm not really using the limiter for sound as much as gain makeup with very light protection.
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Old 12-20-2018, 10:02 AM   #12
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When I'm tracking I don't care what the RMS level is, all I care about is leaving a bit of headroom to avoid clipping. The RMS level of overheads will be very different from fuzz guitar with the a similar amount of headroom.
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Old 12-20-2018, 10:07 AM   #13
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When I'm tracking I don't care what the RMS level is, all I care about is leaving a bit of headroom to avoid clipping. The RMS level of overheads will be very different from fuzz guitar with the a similar amount of headroom.
If recording from an external analog source… (I don't care either btw), if you have it up near zero, you are by definition running the preamp pretty hot by the difference of the level you are using and the sound card's reference. Nothing wrong with that, I do it all the time and on purpose to push my preamp, just a technical aside/good to know.
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Old 12-20-2018, 10:11 AM   #14
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Neither do I but if you have it up near zero, you are by definition running the preamp pretty hot by the difference of the level you are using minus the sound card's reference. Nothing wrong with that, I do it all the time, just a technical aside.
Yeah, I don't record that hot, I always want to leave a healthy amount of headroom to avoid surprises. Drummers can always surprise you though!

I'm only worried about peaks during tracking though, not RMS.
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Old 12-20-2018, 10:17 AM   #15
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Drummers can always surprise you though!
Been there!

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I'm only worried about peaks during tracking though, not RMS.
Yep, same here. The only exception is tracking lots of tracks, if I only pay attention to clipping, there's a chance everything could sum up and start peaking the master pretty quickly once I have them all tracked - then I have to go back and turn them down anyway.

On my really nice preamps, the 4 that ironically don't have output controls, even setting my SC to "low gain" which is even less sensitive, isn't enough to really drive the pre without clipping the DAW. I do have some 25 dB pads for that but don't really use them that much.
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Old 12-20-2018, 10:47 AM   #16
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In the analog world, we obviously can exceed "zero" but in digital, zero is the impassable ceiling.
Somebody on another forum wrote something similar. I know that you know what you mean here, and so did he, but I just feel the wording in both cases might be confusing to some people, so I'm copying my reply from that thread. Much of it has been covered above already, but...
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Please don't confuse 0dBVU with 0dbFS.

In analog, the mark that says 0 is usually the nominal or normal expected operating level. It tells us nothing about the actual limits of the circuit. Most times there is 15-20db above that mark before you hit the rails and start actually clipping. Things often get curvy before that point. Sometimes we like how that sounds.

In digital, the 0 mark is the limit. That's exactly as loud as a fixed point digital file can get, and exactly as far as your converter (ADC or DAC) can get. The digital part is perfectly linear right up to that point and then just suddenly clips off. We usually don't like how that sounds. Sometimes the analog electronics get curvy before the digital part clips off, so sometimes it's not so bad.

This is exactly the reason we have that -18dbFS rule of thumb. Our converters are designed to top out at more or less the same level as the analog gear we attach to it. We said earlier thats usually +15 to +20dbVU, and very often +18dbVU. So the limit of our digital system (0dbFS) is set to the limit of our analog system (+18dbVU) and therefore the nominal analog level of 0dBVU hits digital at -18dbFS. But actually that's just a rule of thumb, and if you want to know how your converter works, you have to look at its specs and usually do some math. They never make it easy. I think my line inputs are calibrated to -20dbFS, but I've seen some with as little as 12db headroom above analog nominal.

Worth mentioning too that the analog VU meters are showing you an average very much like (close enough to) a short-term RMS or LUFS level. It's normal for actual peak levels (the ones that actually matter when we're worried about the limits) to be much higher, and on some sources if you have then right on 0dBVU will still end up distorting or clipping. This confuses a lot of newbs. The digital meters they're looking at are almost always peak levels, but they heard -18dbFS, so they shoot for that and end up quieter than they really need to be. That means nothing in digital really, but it means you're running the analog end of things much lower than it's nominal level and much closer to its noise floor. That's only a problem if it's a problem, but in a lot of cases it can be noticeable. It's complicated quite a bit by the fact that for most of us at home with all in one interfaces, we don't get VU meters. The only meters we have are those in our DAW, which are usually peak based.
But also yes, just turn the knobs til it sounds good. I don't have any preamps worth overdriving, so I just shoot for out of the noise floor but not clipping. In the mix, you just do what you need to do. The way to know if you're hitting a plugin too hard is to listen.

Edit - Actually in some situations where either I don't have a lot of time to dial things in or I don't really trust the source to maintain its level or I feel like I might need to match the recording some other time, I'll just turn my pres all the way down. It's the only place on the knob that I know I can find exactly every time. Well...the other end, too, but that's obviously not appropriate in most cases. Noise is better than clipping most of the time.

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Old 12-20-2018, 10:52 AM   #17
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Somebody on another forum wrote something similar. I know that you know what you mean here, and so did he, but I just feel the wording in both cases might be confusing to some people, so I'm copying my reply from that thread. Much of it has been covered above already, but...
Yep thanks for posting, I'm usually searching for the worded answer that includes some obviousness/aha moment without getting too wordy, then it still seems too wordy for something that is as simple as it sort of is.
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Old 12-20-2018, 10:58 AM   #18
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The way to know if you're hitting a plugin too hard is to listen.
Amen.
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Old 12-20-2018, 11:00 AM   #19
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Amen.
Agree with the reverend.
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Old 12-20-2018, 11:01 AM   #20
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in reaper-it only matters on which recording formats are used really-- try this.
1.make 3 input tracks-all same input.
2.right click+ set each track to a different recording format: eg: 1x 24bit wav - 1x vbr mp3 - 1x wav adpcm.
3.make your source input almost reach 0db on input meters.
4.add same input fx on each track-eg: reaeq-set to +6db of output gain there,so your input signal 'seems to clip on input levels'. < (try much higher clip levels plz,something ridiculous like +48db)

5.record a small passage--making sure 'auto mute' does not engage while recording....
6. try normalizing each file,then null each file against each other.


^ conclusions,or theories from any other users =??
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Old 12-20-2018, 11:02 AM   #21
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If everybody would just read the Yamaha Sound Reinforcement Manual in high school, I'd probably type a lot less.
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Old 12-20-2018, 11:10 AM   #22
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If everybody would just read the Yamaha Sound Reinforcement Manual in high school, I'd probably type a lot less.
Had it since the about 1990 - who knows what I've forgotten since then or what has been added but... I will say it is/should be required reading. I also have the Master Handbook of Acoustics by Alton Everest but that's useless until I revisit it, probably both of them - 1990/91 till now has been a long farking time.
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Old 12-20-2018, 11:10 AM   #23
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If everybody would just read the Yamaha Sound Reinforcement Manual in high school, I'd probably type a lot less.
lolz--why? type more.
that news is ancient history-we are in the now! =)
"
YAMAHA SoUNd REINfORCEMENT HANdbook Part I - Paqe 8-18 It is important for you to know whether the output of a particular piece of equipment is supposed to be matched or bridged, or whether that doesn’t matter. When there is an impedance mismatch (which means the source and load are not right for one another, whether matched or bridged), the results can range from improper frequency response to excess distortion to incorrect operating levels to circuit failure. In terms of specifications, it is important to know what impedances were used when measuring the specs in order for the specs to be reproduc- ible. equipment, many manufacturers have contributed to significant confusion. 8.6. 1.1 Output Impedance EXAMPLE: What is the source impedance of the output in this specification? Output Impedance: 600 ohms If you guessed a 600 ohms,” you may be right. On the other hand, you may be dead wrong! Sometimes, instead of specifying the acutal source impedance of an output, a manufacturer will specify the impedance of the load into which the output is designed to oper- ate. In the above example, it is entirely possible that 600 ohms was not the source impedance, but rather the intended minimum load impedance. © 1987 "

ok m8- you carry on living in 1987--people work with modern tools+equipments..
really,it's >where< we are going though,that matters...that^ matters not!! heh heh heh!
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Old 12-20-2018, 11:20 AM   #24
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ok m8- you carry on living in 1987--people work with modern tools+equipments..
Except the laws physics haven't changed, nor have the underlying principles. Analog audio signals nor sound waves behave any more "modernly" than they did then, and the fact that so many have little idea about what they are advising, which they only learned via internet hearsay, then have the balls to argue it from a stern position of ignorance.
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Old 12-20-2018, 11:27 AM   #25
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and the fact that so many have little idea about what they are advising, which they only learned via internet hearsay, then have the balls to argue it from a stern position of ignorance.
lol-yep- i think your advising something with which your having very little ideas about-
try every step in post#20- and report your actual physical realworld findings> right here,,for all to see--
would you be, ever so ignorant to stay blind to it+ignore suggestions made?
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Old 12-20-2018, 11:34 AM   #26
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lol-yep- i think your advising something with which your having very little ideas about-
try every step in post#20- and report your actual physical realworld findings> right here,,for all to see--
would you be, ever so ignorant to stay blind to it+ignore suggestions made?
I can't even decode what the hell you are saying in post #20 but I do understand how analog vs digital references work.

Just so you know, I made the mistake of clicking on your post as I've had you on ignore since not long after you began posting on this forum (occasionally I'll squint and click view post out of morbid fascination) - because though you have the occasional decent piece of advice (thank you) - everything else is riddled with unintelligible dribble with truly terrible grammar. And I don't mean bad grammar as in being a grammar Nazi, but so bad most people have little or no idea what you are even talking about 90% of the time. I find that interesting as you ignore any possible chance of educating yourself via simple suggestion for all of us to read a book with correct info in it. If you want to successfully participate in a social setting (aka around others), especially a technical one, you're gonna have to up your game a bit.

I typically don't bring ^this up, live and let live and all that. Back to my previous ignore status.
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Old 12-20-2018, 11:40 AM   #27
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Just so you know, I made the mistake of clicking on your post as I've had you on ignore since not long after you began posting on this form

lol- that's fine--you show your hand- i show mine.
do try normal type in plain english,but some is encrypted with large doses of slang+humour=correct
stay blind then-i only see the light that shines within all..including you,old soul.
old souls will soon have a new mission-prepare!!
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Old 12-20-2018, 12:02 PM   #28
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Back to my previous ignore status.
Same mistake here. Let him out, lasted only a few days...
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Old 12-20-2018, 12:31 PM   #29
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Same mistake here. Let him out, lasted only a few days...
Hell, I've considered placing myself on ignore before, I wonder if that works.
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Old 12-20-2018, 01:12 PM   #30
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heh.
server has enabled> *global mute* for all that ignore.


there's little point in following other's mistakes,or triumphs,if not learning from each--
+ @karbomusic> if you use your noddle- your can overdrive your analog gear before it hits reaper for the nice saturations n stuff- then whack a heap of input fx @ recording stage to really hammer home your input tones!!
by recording in float and @ an acceptable sample rate such as 48khz (even better,96khz) =lots of extra gains,cuts+ extra overdriving can be achieved with a hybridized approach..

why follow old rules,regs+other peoples ideas--
for graffiti writers,or many any artists consider this essentially as "biting" > stealing ideas/concepts+methods of doing and then,claiming them methods for themselves....which is quite ridiculous in some senses...
float recording and exporting is the only way togo itb to keep the changes that come about @input+output stages..
*if* - your seriously considering oneself to be any kind of high qaulity only audiophile-then,honestly--i believe this is the route to go.
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Old 12-20-2018, 01:18 PM   #31
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@karbomusic> if you use your noddle- your can overdrive your analog gear before it hits reaper for the nice saturations n stuff- then whack a heap of input fx @ recording stage to really hammer home your input tones!!
by recording in float and @ an acceptable sample rate such as 48khz (even better,96khz) =lots of extra gains,cuts+ extra overdriving can be achieved with a hybridized approach..
Input fx cannot prevent your ADC from clipping, and neither can recording a floating point file.
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Old 12-20-2018, 01:27 PM   #32
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Bri1's posts may not add to technical discussions (then again,maybe they do), but I almost always enjoy reading them.
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Old 12-20-2018, 01:44 PM   #33
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posts may not add to technical discussions (then again,maybe they do)

lol,why thankyou!!
i think you will find,with your own examinations- what is typed goes beyond theory-it becomes practical if actually tried.
try it for yourself-and please post your findings=ty.
does -18db matter to me,in any such way,today? nope.
fear is an illusion that all will deal with as we move forward in this timeline-- the solar energies that may come-will make all see eveything in a much more----hmm--expanded viewpoint!!
some will not deal with all new informations that the light brings-lack of knowledge,sight+senses is what binds each to their own darkness.
the point is--how loud,or how quietly,can any 1 person actully hear!!
it's not signal strengths anymore (although technically,it is)
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Old 12-20-2018, 02:09 PM   #34
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how quietly,can any 1 person actully hear!!
Certainly not below 0 dB SPL @ 1 kHz !!!
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Old 12-20-2018, 02:20 PM   #35
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lolz-- 'ol dave pensado or rick rubin do not give out this info-it's up to each to decide as their choice.
choice is the only thing most humans actually *do* by their own free will-nobody here is ever,ever on their own- each has guides..to...guide and...inspire
all people here-can been guided by unseen masters_ which can take,many,if not all forms.
be guided by experiences,as well by intuitive forces.
why 'listen' to music--> when 1 can become the music.
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Old 12-20-2018, 02:23 PM   #36
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heh.
server has enabled> *global mute* for all that ignore.


there's little point in following other's mistakes,or triumphs,if not learning from each--
+ @karbomusic> if you use your noddle- your can overdrive your analog gear before it hits reaper for the nice saturations n stuff- then whack a heap of input fx @ recording stage to really hammer home your input tones!!
by recording in float and @ an acceptable sample rate such as 48khz (even better,96khz) =lots of extra gains,cuts+ extra overdriving can be achieved with a hybridized approach..

why follow old rules,regs+other peoples ideas--
for graffiti writers,or many any artists consider this essentially as "biting" > stealing ideas/concepts+methods of doing and then,claiming them methods for themselves....which is quite ridiculous in some senses...
float recording and exporting is the only way togo itb to keep the changes that come about @input+output stages..
*if* - your seriously considering oneself to be any kind of high qaulity only audiophile-then,honestly--i believe this is the route to go.
Try some ! Paragraphs are really special !

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Old 12-20-2018, 02:24 PM   #37
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lolz-- 'ol dave pensado or rick rubin do not give out this info-it's up to each to decide as their choice.
choice is the only thing most humans actually *do* by their own free will-nobody here is ever,ever on their own- each has guides..to...guide and...inspire
all people here-can been guided by unseen masters_ which can take,many,if not all forms.
be guided by experiences,as well by intuitive forces.
why 'listen' to music--> when 1 can become the music.
I recognise that critter in your avatar pic, but I can't remember what it's from...
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Old 12-20-2018, 02:37 PM   #38
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I recognise that critter

hahaa- lol-think-paranormal..
another thing to consider is... although 32bit gives superior gain control @ both input+out stages in the reaper box-really-- most people only need around 90db of clean signal to get a great image-
+150db,or into '+200,+300dbdb is loud >obviously,but itb reaper> this is avoiding the dreaded 'clipping' scenario-technically superior @ a cost of playback performances.
if 16bit did not clip-it would be fine..and it is kinda ok until you go over the limit set by the formatting...
the only thing -18 says to me is-- your -18 short of your 'target' nominal signal levels.
if all your tracks are recorded at the full 24bit,or 32bit resolution- you can subtract levels to mix,rather than introduce even more low level noise floor quantizations... > ? no?
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Old 12-20-2018, 02:48 PM   #39
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Input fx cannot prevent your ADC from clipping, and neither can recording a floating point file.
Nope and it was stated at least seven times that -18 dBFS wasn't a hard rule and that most of us don't care - it was actually the underlying point of many, if not all of the replies.

I can also say with certainty that if I crank my A-Designs P1 and there is no reduction of that signal before it hits that ADC, it's going clip and it's going to remain clipped because a ~20 bit ADC != 32 bit floating point file.

Keep an eye on the goalposts, slippery little fellas they are.
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Old 12-20-2018, 03:09 PM   #40
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I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
Klanghelm stuff is made to run at -18db. A lot of Waves stuff too.
When looking for that info in manuals it is often reference to as the: Nominal level. Or normal level of operation.
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