Though it is -18dBFS RMS (300~600ms timeframe). Thus we have 18dBFS "headroom" for the Peaks (they will be usually about -12, -9 or in rare cases -6dBFS) till we clip to 0dBFS (clipping in this case the ADC: analogue to digital converters in the audio interface, which we do not want to happen, despite the fact we can restore such clipped peaks using audio editor such as Audition, Acoustica, RX9, etc!). If you'd like to work by Peaks, then aiming for Peak values at around -9dBFS should be fine in most cases. Inside the DAW we can always "Normalise" values to -18dBFS non-destructively, which should be made to happen anyway during media Explorer browsing, a.k.a. Preview and upon Import, thus we can have relatively consistent perception of the loudness for each audio event we want to preview\import in our project.
Been watching a ton of your videos recently and just want to say THANK YOU. This video especially is one I've been trying to get answered for years and can never get a straight answer.
Always to the point Kenny ... great info for everyone to know about .. for sure tracking in the sweet spot of the preamps is mandatory if you are looking for a gretly sounding mix ...
Hey Kenny! Love your tutorials. Is there a way to automatically remove breathing sounds on a long vocal file? Like you did with ReaFir for background noise. Muting or cutting them manually takes eternity.
If you don't mind shelling out about $30 Waves DeBreath plug in has worked great for me. Pretty much throw it on vocal tracks and works with little to no tweaking
What about electric guitar, Kenny? Some of the plugins I use for distorted tones, claim that their ampsim works best by getting the input signal from the audio interface to be just below clipping. This way the signal is louder than the hissing.
You can raise the volume of the recorded item later after recording process for that purpose and you still maintain the clean recording you can return to. There is really no sense of risking clipping your take, because then you're left with less options.
Good video. Back in the day of recording to multitrack 4" (?) tape wasn't it the case that one should record as high as possible, as close to 0db as possible? Although I haven't recorded to tape in forever, I learned recording to tape and that is stuck in my memory. If true, maybe that's where people get the idea that one should record as hot as possible in digital?
Yep because a lot of people don't understand that digital zero is equal to about + 18 dB on the analog scale. I blame the DAW manufacturers for not setting the red levels around -8 and the yellow levels around negative 12 dBFS.
Ur teaching is sooooo Smooth brother ! I do appreciate ur teaching art ! I understand better than the one with my own language !!! subscribed for sure!
Thanks for this Kenny. Quick question - why wouldn't you use a plug in, such as ReaComp, or even the JS: Volume Adjustment to automatically keep your level at -18dBFS? I can see that if you are recording quite a number of people simultaneously, that would place a strain on computer system resources, but that is the only reason I can think of not to use some form of compression or volume suppression to keep you at or around -18dBFS.
@@REAPERMania Guitar DI input at -18. And then with the guitar plugin we can tweak gain/mids/master & level to make it louder correct? Or in which step do we have to bring the level to -6 db??
Thanks Kenny 😉 ... also for the other two videos! Question: How can I get Meters like yours (... zero on top and -6> -12 etc) ??! Do you have a video where you have already explained this? You can remind me
What if you're recording vocals or guitars with effects? Do you set the volume with the FX disabled then enable the fx before recording? Or do you set the level with FX enabled? Additionally, taking into account that you can "record" the effects live or just "hear" the effects while recording but only record the "dry" signal. Please explain the target level for these two recording options. Hope this makes sense.
I've done exactly what you're saying while trying to record my guitar cab. SM57 panned left and a Sennheiser e602-II panned right on on a mixer from a Vintage 30. My levels are perfect before and during the recording; upon playback it sounds really good but I seem to lose 20db. However, when I record vocals, that doesn't happen. I'm confused.
hello. How about recording electric piano? Which volume level should be set in output/phones of el. piano? I can set output level of piano, and preamp level in audio interface. For example I can set max level at piano and lower level at preamp or minimal level at piano and approperiate level at interface or sth beetween. Which is the "golden level"?
Hi Steve, what do you mean by pull up the mic pre? Do you mean increase the signal from the mic pre? If so, most audio interfaces have a dial or control for each mic pre on the unit itself.
@@Daysofsamara It doesn't even matter how many "bits" are in ADC in your audio interface. Maximum digital value in any ADC corresponds to some maximum voltage on its analog input (signal voltage after preamp). Clipping occurs if analog signal voltage exceeds the maximum allowed ADC voltage, regardless of the bit depth of the ADC.
@@Daysofsamara ADC bit depth only affects the ability to correctly digitize weak signals. For example, if you decrease preamp level to -18 dB you will "steal" 3 bits from signal samples, so for 24bit ADC it's OK, but for 8bits ADC it is catastrophic.
My Go To Channel. Thank you Kenny for putting your time in to help us all out ! Cheers ......
Thanks Kenny 🙏 . One year on and this video keeps on giving to the community.
Mr Gioia, sir, you rock!!!
Though it is -18dBFS RMS (300~600ms timeframe).
Thus we have 18dBFS "headroom" for the Peaks (they will be usually about -12, -9 or in rare cases -6dBFS) till we clip to 0dBFS (clipping in this case the ADC: analogue to digital converters in the audio interface, which we do not want to happen, despite the fact we can restore such clipped peaks using audio editor such as Audition, Acoustica, RX9, etc!).
If you'd like to work by Peaks, then aiming for Peak values at around -9dBFS should be fine in most cases.
Inside the DAW we can always "Normalise" values to -18dBFS non-destructively, which should be made to happen anyway during media Explorer browsing, a.k.a. Preview and upon Import, thus we can have relatively consistent perception of the loudness for each audio event we want to preview\import in our project.
Hi Kenny, I just want to take a moment to thank you for your videos. I have learned a lot over the years and thought it was about time to thank you!
You always have no fluff, useful videos for us Reaper users. I thank you for the guidance,tips and ideas.
Been watching a ton of your videos recently and just want to say THANK YOU. This video especially is one I've been trying to get answered for years and can never get a straight answer.
Thank God I understand English 🙏; otherwise, I wouldn't know what to do without your tutorials 😅 Thank you so much, Kenny 👍🤩
Thanks for your consistency Kenny. You remind me to keep going and keep learning. Appreciate you!
Hey Kenny, I'm glad you remade these videos. Big thanks for all the tutorials!
From a new user of reaper or any DAW, thank you for all the videos!
Always to the point Kenny ... great info for everyone to know about .. for sure tracking in the sweet spot of the preamps is mandatory if you are looking for a gretly sounding mix ...
Hey Kenny! Love your tutorials. Is there a way to automatically remove breathing sounds on a long vocal file? Like you did with ReaFir for background noise. Muting or cutting them manually takes eternity.
You can use Dynamic Spilt but I doubt it would be perfect if the breathing is loud.
If you don't mind shelling out about $30 Waves DeBreath plug in has worked great for me. Pretty much throw it on vocal tracks and works with little to no tweaking
Breathing means you are human.... Well most of the time.
What about electric guitar, Kenny? Some of the plugins I use for distorted tones, claim that their ampsim works best by getting the input signal from the audio interface to be just below clipping. This way the signal is louder than the hissing.
You can raise the volume of the recorded item later after recording process for that purpose and you still maintain the clean recording you can return to. There is really no sense of risking clipping your take, because then you're left with less options.
Good video. Back in the day of recording to multitrack 4" (?) tape wasn't it the case that one should record as high as possible, as close to 0db as possible? Although I haven't recorded to tape in forever, I learned recording to tape and that is stuck in my memory. If true, maybe that's where people get the idea that one should record as hot as possible in digital?
Yep because a lot of people don't understand that digital zero is equal to about + 18 dB on the analog scale. I blame the DAW manufacturers for not setting the red levels around -8 and the yellow levels around negative 12 dBFS.
very well done as usual
This channel is under rated!
Ur teaching is sooooo Smooth brother ! I do appreciate ur teaching art ! I understand better than the one with my own language !!! subscribed for sure!
Thank you Kenny!!!!
Thanks for this Kenny.
Quick question - why wouldn't you use a plug in, such as ReaComp, or even the JS: Volume Adjustment to automatically keep your level at -18dBFS?
I can see that if you are recording quite a number of people simultaneously, that would place a strain on computer system resources, but that is the only reason I can think of not to use some form of compression or volume suppression to keep you at or around -18dBFS.
Thanks Kenny 🍻🥇
thanks kenny
Thank you but i have a question. Above what level can you not mix tracks?
😊 thanks for your help
Should we apply this even when recording through guitar plugins? Thanks!
Your guitar input should still be at -18. Correct.
@@REAPERMania Guitar DI input at -18. And then with the guitar plugin we can tweak gain/mids/master & level to make it louder correct?
Or in which step do we have to bring the level to -6 db??
@@iwillspam5985 Once you're in the plugin you can make it as loud as you want.
@@REAPERMania thanks Kenny. I just wanted to know which db should I aim to for optimal volume/loudness. 0 db?
@@iwillspam5985 this was answered in the video.
Thanks Kenny 😉 ... also for the other two videos! Question: How can I get Meters like yours (... zero on top and -6> -12 etc) ??! Do you have a video where you have already explained this? You can remind me
that's the default scale resolution of the TCP meter in the default version 6 theme if you expand the TCP enough vertically
@@ARE_YOU_SICK_OF_YT_CENSORSHIP thnx Bayan👍🏽
What if you're recording vocals or guitars with effects? Do you set the volume with the FX disabled then enable the fx before recording? Or do you set the level with FX enabled? Additionally, taking into account that you can "record" the effects live or just "hear" the effects while recording but only record the "dry" signal. Please explain the target level for these two recording options. Hope this makes sense.
I've done exactly what you're saying while trying to record my guitar cab. SM57 panned left and a Sennheiser e602-II panned right on on a mixer from a Vintage 30. My levels are perfect before and during the recording; upon playback it sounds really good but I seem to lose 20db. However, when I record vocals, that doesn't happen. I'm confused.
Is the preamp you are referring to external?
Great video, thank you!
hello. How about recording electric piano? Which volume level should be set in output/phones of el. piano? I can set output level of piano, and preamp level in audio interface. For example I can set max level at piano and lower level at preamp or minimal level at piano and approperiate level at interface or sth beetween. Which is the "golden level"?
Anywhere between -18 and -12 should be fine.
Always good to go over this stuff again. Thanks! 🎙🎚
Thanks Kenny! I have read digital is not analog in any way. Too hot is not what you want in the digital world.
How do you pull up the mic pre for each track when seeting recording levels?
Hi Steve, what do you mean by pull up the mic pre? Do you mean increase the signal from the mic pre? If so, most audio interfaces have a dial or control for each mic pre on the unit itself.
Hi!
I cannot have the proper level without background noise. How can I fix that beside noise reducer?
Great video and recommendations! Do you recommend aiming for -18db in every case?
Yes, absolutely
@@REAPERMania Even for an electric guitar recorded directly using an audio interface with amp simulators? Excellent channel, by the way.
@@jaortizco I would, don't see a reason not to.
@@jaortizco Sure
@@David_Logr Correct
Cool!!!!!**** Philippe Bouchard!!!!!!********
To summarize this video:
-18db
R u sure? Question:
Peaks, RMS or LUFS?
@@PASHKULI the default REAPER setting is multichannel peaks and since Kenny didn't change that, it's safe to assume that this is what he used
@@ARE_YOU_SICK_OF_YT_CENSORSHIP then it would be wrong
@@ARE_YOU_SICK_OF_YT_CENSORSHIP Correct
@@PASHKULI How so?
Yeahh all set to -18 simple!
why Reaper is the only daw without input faders?
First.... And thanks for this. Very helpful
Glad it helped!
Your mom is a proper recording level.
Hey, Kenny! New sponsors? :-)))))) Seriously: record in 32bit float wavpack and don't worry about clipping never ever again.
Your interface ADC is still in 24bit integer, so even if you record 64 bit float, the signal is clipping in the ADC before reaper Input.
@@Daysofsamara yeah
Kenny actually already did a video on this a while ago.
@@Daysofsamara It doesn't even matter how many "bits" are in ADC in your audio interface. Maximum digital value in any ADC corresponds to some maximum voltage on its analog input (signal voltage after preamp). Clipping occurs if analog signal voltage exceeds the maximum allowed ADC voltage, regardless of the bit depth of the ADC.
@@Daysofsamara ADC bit depth only affects the ability to correctly digitize weak signals. For example, if you decrease preamp level to -18 dB you will "steal" 3 bits from signal samples, so for 24bit ADC it's OK, but for 8bits ADC it is catastrophic.