VU Meter

Just recently I had some long discussions with a Cubase using friend about Gain Staging. He suggested that I purchase PSP’s Triple Meyer and start using the VU meter on the first insert slot in an input channel. The idea being that I can make sure that the incoming signal is at a proper level and not clipping. If the signal average 0 on the VU Meyer it will be about -10db.

The diffence this has made to my guitar and bass sound has been extraordinary. I have always struggled to get clarity in the bottom end. My bass guitar was often unpleasant to listen to in isolation. (And not just the playing.) Using the VU Meter meant that it was really easy to get a consistent and well regulated signal. I have been knocked out by the punch and clarity that the bass now has.

The clean sounds with my guitar have improved beyond all expectation. The issue I had sometimes with harshness has gone. For the first time I don’t feel that I am continually having to tailor my sound and alter settings. The overdrive sounds are also much improved in that it is now easier to tell the difference between different amps and overdrive pedals.

I now apply this approach to any incoming audio. I used to think I was careful about levels - obviously not. I am sure that many of you have experienced this. If you haven’t, try it it might just change your musical life.

PSPs meters are great and your friend’s suggestion is valid. However, you could achieve the same result by just adjusting your gain accordingly in a DAW environment versus an analog one. Nominal peak in a digital environment is -12dBFS so I try and get my input signal between -12 and -18 so that any buss summing or plugin/effect gain added after the fact doesn’t have me pulling down the fader in the mix constantly.

That is true enough. However I found the VU meter very easy to work with. It is also large enough to be difficult to avoid so that you are aware of any gain transgressions. I am sure that many here are “au fait” with the whole concept of gain staging. I thought I was until I started being fairly strict about it and realised how many problems I was making for myself.

That’s been the issue with most DAWs as they use sample-peak metering rather than true peak. Even though you can customize the meters in Cubase to deal with this, I don’t understand why better channel metering, especially within the insert buss, isn’t an option. Version 8 does have a new loudness meter that does incorporate true peak levels but it’s on the master buss. Both PSP and Zplane make good metering tools.

Perhaps we ought to request some accurate and assignable meters for the input channels? Something that shows the peak values and the average values.

This sounds like cute placebo effect. How can metering change the sound in any way? It can not.

What about using the Stereo Channel plugin included in Cubase?

My mistake the plugin I thought was included is actually from SleepyTime Records that I installed earlier in Cubase 7 which automaticall migrated to Cubase 8
http://sleepytimedsp.com/downloads

It would be great if Cubase had an rms meter mode - long wanted that

No it wasn’t a placebo effect, just that I was recording the guitars a little too hot. Clean sounds really are better.

http://www.soundonsound.com/sos/sep13/articles/level-headed.htm

See what you think of this article.

A previous thread:

http://www.steinberg.net/forums/viewtopic.php?t=57467

Spend some time around a mastering engineer and then see what you think.

OP is talking about levels while tracking. Peak meters are more helpful in such situation, because they show you peaks of your signal. RMS meter doesn’t tell you if you are clipping the signal or not. If the OP’s problem was clipping while tracking with peak meters, then then RMS meters are surely not the solution to the problem. In the end, peak or RMS, makes no change to the signal, signal is the same, different meters are just measuring that same signal in a different ways.

I can only share my experiences. I personally find the VU meter helpful and the sonic results seem to bear that out. Peak meters do indeed show clipping but do not show that a signal is too hot. If you were recording on a tape machine this would not matter, indeed the resulting warmth would be desirable.

If you have a signal that has been limited to 0 dB it will still peak because error correction circuitry in CD players will reconstitute those peaks. The whole point is to get you head around a significant lowering of the incoming signal. It is also important that the outgoing signal is treated the same way. This then allows the mastering a better chance to do what is needed.

I don’t think that the loudness wars have helped as people have got used a more unnatural sound.

You don’t have to agree with me. I can only suggest you try it. I am not going back to my old habits.

Certainly you should keep levels well below clipping. But that which is suggested in your above mentioned SOS article, to keep all peaks between -18 and -12 dBFS, that is not practical at all. It makes editing hard, levels are too low to see. Then you have to boost global clip waveform magnification, but then all commercial loops and clips are way too hot, and look like clipping. So what is the author of SOS article proposing as solution? To batch process all files and reduce levels to -12dB peak! This is not practical at all, you simply cannot bounce every file that you import, just to have it 12dB lower.

To estimate good levels, you can use one of Waves signature plugins, which all have input LED that shows you best input operating level - LED must be yellow to orange. You will see that they need quite hot signal to work optimally. For drums most seem to need peaks at -6dB to match default sensitivity. It depends on audio material, so it must be adjusted properly for each track.

What I do is record audio to peak around -6db or so. I go for as high as possible, but not to the edge. This gives me good view of waveforms for later editing. Then I keep this level to feed insert plugins. But just before hitting fader I adjust level with last plugin in chain (or with limiter in channel strip), so that my fader on remote control is around its sweet spot 0dB, which gives me most resolution while mixing. And I keep master bus level at -10dB, which gives me enough headroom.

Anyway, find your own best practice, and work on it.

Thank you for your thoughts on this. I agree that proprietary samples are often too hot. However, that in itself does not negate the basic principle or indeed make the way the samples are produced right. I certainly don’t find that the wave forms are too small th edit properly. Possibly they are two thirds of the size they were. Not too much of a hardship. My use of samples is generally limited to Kontakt. As a guitar player I am more concerned with how my guitar or bass sounds. Actually it is the bass sounds that have improved the most. Anything that improves the sound in that direction has got my vote.

You will work the way you want to, as indeed we all do. I am just, in a traditional forum way, sharing my experiences. I am not claiming any expert knowledge in this field. I must say that I respect the guys at SOS as people who have real experience in this field.

Remember that your front-end, analogue pre-amps etc. are more likely to sound better at 0dBVU.

If you track your bass through the analogue chain (xxx) at a too high level, it is here you are muddying/distorting your bass signal. Lower this to follow the “dancing” around 0VU, will clean up your signal (best S/N ratio and best/cleanest signal when it comes to optimal level according to distortion).

The gain staging is more important to the analogue stage, and here is the biggest misunderstanding :wink:

The other benefit (besides cleaner/better signal) in using 0VU at input, is that you don’t have to worry about clipping your signal in the digital domain. You have gotten yourself proper headroom.
The peak value will vary according to sound source/signal. Drums will peak higher than a full on distored guitar amp. So…

So follow the 0VU on input, and the peaks will end up where it’s normal (but different) for type of sound source.
The same peak level for all different signals is just plain wrong, and if you follow good old gain staging it is 100% normal that the peak level will be different

Remember!
The most important part of the gain staging is in the analogue front-end.
And also remember that the VU-Meter is not fast enough to let it read 0VU for fast transient sources like drums (you will likely have it to read -7 to -5 dBVU). So don’t forget the peak meter completely.

My advice I use to give, is to peak transient rich signals (drums, perc etc…) to -6 to -10dBFS, and long sustaining sounds (synth pads, distorted guitars etc…) to -12 to -18dBFS.
This is a general advice, but are very close to equal the healthy 0dBVU through your analogue chain.

Added noise/distortion through your analogue front-end will follow you through the project from start to end, so please take your time to get the signal right from the start :wink:

PS. For editing there are usually a way to adjust the size/readability of the files (without messing with level).

Thanks for your input Piirka. Recording the sound too high(0db) was my problem in the first place. It was why I used the VU meter as it is calibrated to read a signal of 0db at -10db when used in the first input vst slot. Yes I know that it is not as sensitive but it is much easier to get a reasonable idea of what is happening. Actually the digital VU meters are much more responsive than the real thing.

I am finding that the optimum level for my bass is -18 to -10 db. Once sound is in the digital domain then I do agree that there is not so much of a problem. You can’t run a sound level that is ok at 0db in analogue through the analogue to digital conversion. It will clip the digital signal.

You can’t run a sound level that is ok at 0db in analogue through the analogue to digital conversion. It will clip the digital signal.

Note that Piirka specified 0dbVU

fs (full scale) is what you see in your digital meter and yes this of course would clip…no-one is claiming otherwise.
What you seem to to be missing is that -18dbfs is a popular (industry standard?) level to calibrate to 0dbVU.

So working at 0dbVu in analogue is actually perfect for then recording to your AD. It will be -18dbFS.

You are quite right. I need new glasses. I don’t know why I missed that. Especially as Piirka was talking such sense.

I thought the PSP VU was set to -10db. I am happy to stand corrected.

I am surprised that this subject is not one that comes up more often as it seems to me to be such an important and basic staring point in the recording chain. I feel such an idiot for not getting to grips with this earlier.

No need to feel that way at all. So many DAW users these days have never laid their hands on a real console and experienced what true gain-staging is all about. And, because the “louder=better” crowd is still alive and well, you still have “experts” walking around spouting the myth that if you don’t get your signal just under clipping, you’re getter a weaker recording. As I said earlier, the PSP meters are great but don’t do anything that you can’t already see in Cubase and following the information in your SOS link will always render positive results.