Like Karbo said, it's just a "reference level" that some kind soul decided was a good idea for modern daw users to know since the vast majority of them never worked in analog and for the most part their prosumer audio devices have shitty meters. It's something to aim for if you don't know what to aim for or you don't trust your ears.
For the other thing, I'd bet a dollar to a doughnut that with an isolated audio track with Slate Tape or whatever other analog modeling thing on it, very few people would be able to actually reliably hear the sweet spot they obsess over in the blind. That I could feed that thing -25 to +10 (and equalize the output gain) and they'd only really hear the distortion or difference on the high extremes when I pushed it really hard. I'd even go so far as to say a certain percentage wouldn't even know (in the blind) if the plugin was bypassed of not.
For the other thing, I'd bet a dollar to a doughnut that with an isolated audio track with Slate Tape or whatever other analog modeling thing on it, very few people would be able to actually reliably hear the sweet spot they obsess over in the blind. That I could feed that thing -25 to +10 (and equalize the output gain) and they'd only really hear the distortion or difference on the high extremes when I pushed it really hard. I'd even go so far as to say a certain percentage wouldn't even know (in the blind) if the plugin was bypassed of not.
^hmmm-yuz--now that's a very interesting reply=thanx!
What I found while doing some very deep extrapolation was that if you put good sounding stuff into the input side of many FX, what comes out the exhaust pipe side is also good.
__________________
Glennbo
Hear My Music - Click Me!!!
--
For the other thing, I'd bet a dollar to a doughnut that with an isolated audio track with Slate Tape or whatever other analog modeling thing on it, very few people would be able to actually reliably hear the sweet spot they obsess over in the blind. That I could feed that thing -25 to +10 (and equalize the output gain) and they'd only really hear the distortion or difference on the high extremes when I pushed it really hard. I'd even go so far as to say a certain percentage wouldn't even know (in the blind) if the plugin was bypassed of not.
I never thought of such thing as a "sweet spot" in plugins.
Maybe it is valid for analogue equipment... like pre-amps, amps, compressors and alike.
Plugins work with numbers. Most of the time (99%) I am not able to hear the peaks hitting the compressor (most of them have a duration below 10ms, it is impossible to be heard) - fast Attack, fast Release and hope you are done. Checking the wave peaks on the sample-meter waveform won't hurt, as it shows you what peaks have been altered (Ratio X:1).
Some people after recording has been done do a Normalisation. Some do it at 0dBFS, some prefer -3dBFS, others -6dBFS.
But if you don't do the Gain staging as a next step, you are going to have a hard time with the faders.
I never thought of such thing as a "sweet spot" in plugins.
^heh- there's a newbie section for these type of thoughts- if you cannot read the plugin input/parameter+output numbers-your fooked.. use ears ,aye.
as for normalizing-it makes sense here to use it+after that process, using the max limit of clip/item gain reaper provides..is that 24db? or 24. somminsommin.. heh.
the recording levels of the items,or clips,will determine what level hits the plugins if using either take fx,input fx,or insert fx,master fx--otherwise,floating point takes care of clipping pre fader in reaper-- it's only the post item/clip gains that matter @ the faders+master outputs. right?
^this is where the 32bit internal rendering comes in handy-- because- then you can start to decide which type of gain you would consider the most pristine,or clearest signal boosting method... pre amp analog boosting,or digital bit boosting.... right?
airwindows made a plug he called purist gain >i think< -- is this preferable to pre amp gains,or wut?
On the contrary, I look at things "realistically". I know that the overwhelming majority of the target audience of those kinds of things, which are pretty subtle by nature unless you drive them, can't really hear them.
That's why some obsess over the -18 level hitting them, because they can't actually really hear it, but more just trust that's where the good sound is.
That's not a knock on anyone. More to say that it's a consumer market and most consumers of those products don't have the critical listening skills or monitoring environment to hear some really subtle things.
And as Pipeline will probably tell you, often enough hitting analog stuff hard(er) or driving it actually sounds better so if you're sticking to the -18 rule for your analog plugins you might be missing something. It might actually sound better at +5 analog or -13, depending on the source.
And as Pipeline will probably tell you, often enough hitting analog stuff hard or driving it actually sounds better so if you're sticking to the -18 rule for your analog plugins you might be missing something. It might actually sound better at +5 analog or -13, depending on the source.
aye-some good points considered--
totally think all metering and plugins are frequency + transient energy dependant--source materials really vary in rms<>peaks,so as voltage is a continous wave in theory-the fluctuations happen as rapid the frequencies going in..eh--
^bearing that in mind- is why i previously suggested manufacturers devising a variable sampling system--as nutty it may seem -it could make computational logic-- matching analog input frequencies to sample frequency rates>>not all instruments produce the amount of cycles we set our interfacing at--it's the reason why at a fixed sampling rate 1 cannot have truer (fm) freqmodulation--rates need to be variable,like voltage does for that to really happen itb...anyways veering off topix...
questions still remain-- like_ "which type of input and output gains are preferable these days while using reaper: digital,or,analog?
That's why some obsess over the -18 level hitting them, because they can't actually really hear it, but more just trust that's where the good sound is.
Pretty much ^this. Why it is that number... and chasing that number, are two entirely different things. It's more a "nominal" result than a target when incoming from the outside world. I have to wonder if there were ever a scenario where -18 dBFS RMS sounded like shit (proverbially), but someone kept it there because they were told that's the place it should be LOL.
__________________ Music is what feelings sound like.
I have to wonder if there were ever a scenario where -18 dBFS RMS sounded like shit (proverbially), but someone kept it there because they were told that's the place it should be LOL.
Probably not, unless the signal was just shit to begin with. The "acceptable range" of 24-bit digital (and 32-bit float or greater internals) is so wide that it kinda doesn't matter unless you clip or record a noisy signal at -50 or something.
It matters more on the analog front end as relates to noise and distortion and S/N ratio and all that. But you know that obviously so... preaching to the choir.
yaaay let's see what 'ol chris has to say about floating point math+sampling rates+gains with the bitshiftgain vid> (and why it may be usefull to some)
Probably not, unless the signal was just shit to begin with. ... But you know that obviously so... preaching to the choir.
Yea, I simply meant someone being so concerned they proverbially might ignore their ears because someone told them "this number sounds best", I had to invent a theoretical "sounding like shit" scenario to make the point, hope that makes sense. Not real scenario, just identifying the potential absurdity of worrying about it.
__________________ Music is what feelings sound like.
Yep. Anyway, I do have some console emulation things like the CTC-1. What I do is drive it until the sound is really obvious, then start backing it down until I like what I hear, and then A/B it on/off a few times to maybe get a good impression of what it's doing to the sound, paying no attention at all to what the input level is.
questions still remain-- like_ "which type of input and output gains are preferable these days while using reaper: digital,or,analog?
Bri1, as I wrote above, some of us would love to be able to record at 32bit/384kHz, why not record even at 64bit (which will catch every ~2dB per 1bit, how cool would that be, right?).
With that said, imagine your files sizes.
Now consider that most people (even musicians) can not tell the difference between a CD (44.1kHz/16bit) and mp3 (320kbps).
With that said, I think that the "sweet spot" here is 24bit/96kHz for recording, mixing (internally In the Box it can spread to 64bit but the samplerate should be the same 96kHz), mastering audio.
Me, personally I record at 24/48kHz (I can do it 24/96 but the file sizes worries me a lot, and with backups, renderings... too much)
Something tells me that even the best audio engineers won't be able to tell the difference between 24/96 and 32/384 | 64/768 on a blind test, spot on 10 out of 10 times.
some of us would love to be able to record at 32bit/384kHz, why not record even at 64bit (which will catch every ~2dB per 1bit, how cool would that be, right?).
With that said, imagine your files sizes.
So, again - the sweet spot.
heh-you know what m8--in a perfect bubble we would all have them rates as a bare minimum!!
people should expect it as nature gives it..
concerning files sizes--again,this is where wavpack format compressions can come into play-
also with windows,there is a format available in reaper -called adpcm--this will squash 64/32 bit down to 4bits,which i think gets expanded back to 16bit on file details..not sure if that includes a dither in that process tbh.
if we wanted to try and recreate some of natures sound levels-64bit still is not enough really..cosmic radiation+the earths core generates mega levels that would be insane to push them levels back via speakers...so your right about 'sweet spots' as we do not want to actually damage ourselves while playing back any type of sound--such as a hand grenade @ 2 paces.. might
maybe if more people done more blind tests with higher qaulity calibrated systems--the results may differ also.
-18db? ...with plugs like bitshift,or any other pre,or post gainer itb--- means little other than the pre determined expected plug input levels...usually miscallibrated for 32 or 64 bit fwiw...
it totally matters if you try to record that itb @16 or 24bit as they are fixed point..clippers.
reaper's meters only go down to 24bit levels anyways,as far i tell= -150db.
It is up to you how you would like to mix. Some people use Normalisation on every file, load compressors, and bring the faders down.
I prefer the most aggressive peaks (of transients) to hit around -6dBFS, before applying FX like compressors, EQ, spacial...
I do not pay much attention to the RMS (as it varies a lot, depending on the "time window" settings) before pre-master.
So, -18dBFS matters when you want to have a safe reference point (safe for the Peaks and True Peaks not to clip accidentally). It is up to you how and when to treat them with FX (compressors usually).
Some people pass them through analogue compressors and bring that output back In The Box (ADC → computer).
Digital domain is less forgiving, regarding Peaks (if forgiving at all actually)
Loudness is different and it is up to the mastering work to deal with it.
Nowadays people tend to use more and more 'pre-master FX chain' on the Master bus: usually a compressor, saturator, master EQ and limiter. And mix with those settings as a final output. If the Master bus clips, either change some levels, or the FX params. on the Master bus or individual tracks.
At the end of the day, important is what's coming out from the speakers.
32 or 64bit itb recording gives us all great resolutions,without the input or output clipping scenerio,so,why not use them to full capacities for maximized processes for clarities sake-- ?
cd was always arguably shyte clarity wise -but- commerically,it made perfect sense to resell back catalogs at a set price v manufacture material time+costs.
w/e. floats that bloat.^
if any user values their songs or recordings in any way- they would also care about maintaining the highest possible signalling chains they can afford--but in the pocket+computationally...
currently computers and filing systems will choke at these much higher rates,but the technical details say the math is better for clarities sake overall..
whatever rates people want to then murder their audio with after the full 64bit precisions are done-- then that's an artist choice as well as streaming services,or media playback devices,matters.
so long any users get back the sound levels any modern daw provides,with the specified levels according to interfacing manufacturers frequency response charts--all is well..right?
the snr of various audio interfaces,mics+instruments going direct-differs greatly so input levels can really matter itb,but only really for 16,24bit input recordings.
^a 16bit mic input going into a 64bit recording chain..seems to lack the db levels required to fill the scale..unless boosted,somewhere.
lolz@da trollz.
some call it science-a demonstration of measurable,repeatable results some regard as 'fact'.
see transformers not only appear in the movies >
heya Coachz-um,as you see this as your thread (that some wannabe nerd apparently jacked) >
would you mind being a bit of a case example ?
i mean,would you be willing to provide your realworld specs-- mainly your recording process..
can you provide your specific audio interface spec sheet+ what ever other microphones,guitars,or other analog keyboard specs you currently use to record with??
or,do you work 100% itb,all vst/sample libraries??
what's your particular recording chain,today?
with these numbers 1 can do some calculations to see where you might gain,or lose audio informations..
32 or 64bit itb recording gives us all great resolutions, without the input or output clipping scenario, so why not use them to full capacities for maximised processes for clarity sake?
I thought there is hope for you, Bri1.
But you still regress back to your comfortable ignorant bliss.
Why not use 64bit so we can record even jet engines directly with mics behind the turbines? No signal clipping guaranteed! Not sure about your mic though... might need a spare one afterwards.
Even better, with 64 bit recording we could take in the voltage output signal directly from the valves of the amps! No clipping guaranteed. Nice valve sound, without all the heavy stuff. Then we can treat it with plugins (Impulse responses and Cabinet plugins... we might need a new audio interface or computer).
heya Coachz-um,as you see this as your thread (that some wannabe nerd apparently jacked) >
would you mind being a bit of a case example ?
i mean,would you be willing to provide your realworld specs-- mainly your recording process..
can you provide your specific audio interface spec sheet+ what ever other microphones,guitars,or other analog keyboard specs you currently use to record with??
or,do you work 100% itb,all vst/sample libraries??
what's your particular recording chain,today?
with these numbers 1 can do some calculations to see where you might gain,or lose audio informations..
plz+ty.
I have a 980x cpu, 12 gb ram, GA-X58A-UD3R motherboard, bequiet case, 3 hard drives and an rme 9652 interface with 1 digimax fs 8 io over fiber. I have Rokit 8 monitors with a 10" sub and a 27" display, 2 trion 8000 tube mics, a shure sm7b, 2 blue 100 mics and 2 karma audio pencil mics, 6 string seagull acoustic, 12 string seagull acoustic, line 6 variax, Agile Les Paul, Ibanez attack bass, tonelab le pedal, Tech 21 bass VT pedal, acoustic 20 watt amp, dsl15c amp, dsl40cr amp tomorrow if all goes right and presonus hp4 headphone amp, behringer UMX610 controller, Kawai ES8 keyboard, mandolin, alto saxophone, cordobo gypsy kings guitar and that's about it. It seems to work best at 128 buffer and I have to disable cdroms and malwarebytes file protection to not have latency mon issues. I record at 44.1 16 bit and sounds great to me.
I had a lot more but a tornado visited me in 2015 so I had to start over.
heya-o that's great info-thanx for the spec sheets.. glad your happy !
question still remains though.. when does 18db matter> to you,right now using reaper recording @16bit?
oh + da bombshell-- do you think fibre optics was considered when people calibrated their plug in da box?
sound does not need to be carried via the air- it can travel as light also..it's just our ears are kinda made for that purpose specifically..the whole body reacts to the vibes though eh.. @what level,does them vibrations cease? -inf db,or -180db?
many questions...
heya-o that's great info-thanx for the spec sheets.. glad your happy !
question still remains though.. when does 18db matter> to you,right now using reaper recording @16bit?
oh + da bombshell-- do you think fibre optics was considered when people calibrated their plug in da box?
sound does not need to be carried via the air- it can travel as light also..it's just our ears are kinda made for that purpose specifically..the whole body reacts to the vibes though eh.. @what level,does them vibrations cease? -inf db,or -180db?
many questions...
Edit: Carvin should be Carver
The designer of Carver amps said that if he had two amps and the differences at any sound at any frequency were more than 40db down, humans could not tell the difference. He would get a high end amp like a McIntosh and tie the speaker negative to his amp and the positive from the McIntosh and his amp to a speaker's +/- and any differences in the transfer function would come out as sound. When he got them 40db down compared to the input he said the amps were effectively the same. He successfully duplicated $12,000 amplifiers with equipment he would eventually sell for $600-$1200 and passed blind listening tests by both magazines.
The designer of Carvin amps said that if he had two amps and the differences at any sound at any frequency were more than 40db down, humans could not tell the difference. He would get a high end amp like a McIntosh and tie the speaker negative to his amp and the positive from the McIntosh and his amp to a speaker's +/- and any differences in the transfer function would come out as sound. When he got them 40db down compared to the input he said the amps were effectively the same.
Utter BS.
Just watch the video, and stop making excuses for your ignorance:
So you are disputing what ? How Carvin modeled his amplifiers ?
Carver caused a stir in the industry in the mid-1980s when he challenged two high-end audio magazines to give him any audio amplifier at any price, and he’d duplicate its sound in one of his lower cost (and usually much more powerful) designs. Two magazines accepted the challenge.
First, The Audio Critic chose a Mark Levinson ML-2 which Bob acoustically copied (transfer function duplication) and sold as his M1.5t amplifier (the “t” stood for transfer function modified).
In 1985, Stereophile magazine challenged Bob to copy a Conrad-Johnson Premier Four (the make and model was not named then, but revealed later) amplifier at their offices in New Mexico within 48 hours. The Conrad Johnson amplifiers were one of the most highly regarded amplifiers of the day, costing in excess of $6,000 a pair.
It might actually sound better at +5 analog or -13, depending on the source.
Linear Mode
