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Input level for recording a bass


Mornats
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I'm still on my quest to record a good bass sound and I think I've found something that helps. When recording I seem to lose all semblance of tone from the bass and if I manage to get some of it back, it's mainly from spending hours and hours tweaking the EQ.

I usually record with the input level as high as I can get it without it clipping but any bass amp simulator I put on that never sounds good or clean. So I tried lowering the input level and amplifying the bass with Ampeg SVX which does sound a bit better.

I'd give you audio samples to compare but I actually used two different basses so it's not a fair comparison but here's how the wave lines look. The top one is with a high input level and the bottom one is the low input level (that I think sounds a bit better when amped with the amp sim).

So is it better to record low and amp up in software or to record high and just EQ the bass? I fear that I'm losing some dynamic range by recording low and that amping it up will amplify any noise/artifacts.

[attachment=153957:bass-wav-lines.jpg]

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How do you plug your bass in? Do you use the instrument or line level input? I practice the bass plugged into a computer and never had this problem but there are a few things you need to keep in mind. Tell us more about your setup.

Edited by kyboo
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I plug it into my Guitar Rig Session interface. I'll select instrument level for passive basses and line for active ones.

That video was brilliant for both comedy value and simple advice. I tend to put a high pass filter on my bass though. In fact, here's the EQ I usually use for my Overwater Classic J which looks like I'm trying to overcompensate for something that I'm not doing right:

[attachment=153962:bass-eq-classic-j.jpg]

That EQ was used for the bass on this track if that helps: https://soundcloud.com/beautiful-skin/gonna-run

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If I put such EQ on my bass signal it would not sound good. However, I don't mix the bass signal with other instruments I just practice alone so I don't know much about mixing in the context of other instruments. Anyway it seems to me that cutting it deep around 300Hz is not a good idea as that is the main area of the bass sound. At least to my ears. How about not cutting there and maybe not using a high pass filter as well?

I use Reaper as well, let me upload my project file.

Edited by kyboo
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Also, I would not use line level for active basses. It is not that much about 'strong' signal but about impedance matching. Not many active pre-amps have the correct impedance for line level inputs. You get more noise if it is mismatched.

Edited by kyboo
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Nice one, thanks for that! It's a much simpler curve on the EQ than mine and it works (a little cut around 500hz cuts out some mud I find). My problem is reading too much about what to cut and boost and not listening to what I'm recording enough.

Oddly enough it's the bass that I still have trouble recording/EQing/mixing despite being a bass player.

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Glad to help. I find a compressor having much more impact on a good bass sound than EQ.

Also, have you tried RedWirez bass cabinets impulse responses? They are amazing http://www.redwirez.com/bigbox.jsp

Lastly, this analog tape simulator is worth to check out as well http://www.masseyplugins.com/#plugins/tapehead Free version does not expire but it does not remember settings between restarts. It has only two knobs so it is not much of a problem. It sounds nicer than any tube simulator I tried.

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This could get quite complicated a boring very quickly.
I also use the IK bass rig and the trick is not to overload the input.
Do you use 24 or 16 bit?. I use 24 bit has it allows you more headroom, I try to hit the meters around - 12db sometimes less.
Good gain staging is important, understanding it will improve how you mix. When the signal has gone through SXV and all the processors it should hit the master buss roughly the same has it went in, look up Fletcher Munson. We preserve louder to be better so to make good judgements you need a similar volume.
The recording level of the DAW should not make any difference but how you drive the AD converters will especially on the cheaper soundcards try not to overdrive them.
I try to get the sound right has early has possible and only use EQ to help the sound fit in the mix.
To recap use 24 bit and try to hit the meters at -12 to -16db when the signal has gone through SXV and all the processors it should it the master buss roughly the same has it went in.
Just to contradict myself all amps preamps and some plug- ins have a sweet spot, the trick is to find it, if the signal is too hot try to bring it down before it hits the convertors

Edited by ironside1966
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  • 2 weeks later...

Tongue in cheek or otherwise, the alarming thing here is that I've worked with engineers who have done exactly this, so I will raise your ;) with a :o! The main issue is the nuances of dirt/compression ultimately get lost in the final mix...you're just adding another level of dirt behind the already distorted guitar(s). In this example, what you're hearing is the phat bass frequencies, which you can achieve to a degree by tweaking the EQ on the individual track.
P

[quote name='kyboo' timestamp='1391364011' post='2356244']
This video might help ;) [media]http://www.youtube.com/watch?v=mol6awk_aDo[/media]
[/quote]

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[quote name='ironside1966' timestamp='1391376733' post='2356542'] This could get quite complicated a boring very quickly. I also use the IK bass rig and the trick is not to overload the input. Do you use 24 or 16 bit?. I use 24 bit has it allows you more headroom I try to hit the meters around - 12db sometimes less. Good gain staging is important it will improve how you mix, when the signal has gone through SXV and all the processors it should it the master buss roughly the same has it went in, look up Fletcher Munson. We preserve louder to be better so to make good judgements you need a similar volume . The recording level of the DAW should not make any difference but how you drive the AD converters will especially on the cheaper soundcards try not to overdrive them. I try to get the sound right has early has possible and only use EQ to help the sound fit in the mix. To recap use 24 bit and try to hit the meters at -12 to -16 db when the signal has gone through SXV and all the processors it should it the master buss roughly the same has it went in. Just to contradict myself all amps preamps and some plug- ins have a sweet spot the trick is to find it if the signal is too hot try to bring it down before it hits the convertors [/quote]

This.

Always plug an instrument into an instrument in, not a line in, unless you have an RME UCX or equivalent which figures out the best impedance to match against the connected device with the kind of cold efficiency you would expect from a class A German product

Recording at a high level is a very bad and unnecessary thing to do in the digital realm, so just dont do it.

Record at 24bit, the digital noise floor is then super low and will not adversely affect your recording.

Give yourself a good 12dB of headroom above your loudest peak. You can always use the free GGain vst to add some level later if required, you cant capture a transient peak more accurately if it goes too high on the way in.

Do pay attention to your analogue gain staging on anything you are runnung into before the interface, get that at its sweet spot, my Audient mic pres like to be driven a touch harder than my RME mic pres, but sound awesome either way in truth.

Get the sound as cool as can before you hit tape, however there is nothing wrong with the eq you have there if its like that to fit into a mix, Repaer's EQ is so powerful that you can get far more surgical than that if you need to, it certainly doesnt look overly complex to me.

On the other hand if you are doing that to make it sound great to your ears on its own out of a mix then you can probably get closer to your ideal tone out of the box IMO.

When making decisions always always match the levels of changed vs original or your perception will be squewed toward the louder signal, every time!

Edited by 51m0n
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Some brilliant advice, thanks everyone. I know where I'm going wrong. I'm plugging my bass into my Guitar Rig Session interface and turning up the input gain until it's just below clipping. In Reaper, I'm recording at around -3 to -1db. (I record mainly with passive basses so I'm pretty much always using the instrument in not the line in.)

I usually start a track with drums from EZDrummer. If I go for a high velocity on the midi files the drum sound, without any EQ or compression can play back at around -0.5db. So this is what I'm trying to match my input signal to when I'm recording my bass. So this is obviously too high and I need to be pulling down the sliders on that drum track to allow me to record the bass at the -12 to -16db that's being recommended.

Gain staging is something that I've not been doing at all. I record quite high and shove t-racks on the master channel to get it to around the same loudness as what I can hear from commercial MP3s on my PC.

I need to do more homework about the various staging of "volume" (i.e. gain staging, what level to have my sliders in Reaper, when to and how to increase the overall volume at the end so it's as loud as a nicely done commercial release).

This is what the mixer window looks like for this track which may give an indication to what I'm doing wrong (or right?!): [url="https://soundcloud.com/beautiful-skin/beautiful-skin"]https://soundcloud.c.../beautiful-skin[/url]

[attachment=155073:mix.jpg]

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Where can input trim come into play ? (On the DAW channel)
The channel fader lowers what's going to the master Buss (or a Buss) etc..
So if you have an audio file that is quite hot (not over 0db of course)
would the thing to do, be to pull back the input trim ?
I tend to do this if I am going to use a few effects on that track, and I need some spare headroom.
Would this be the correct way of thinking ?
I know it's better to have an original file that's not so hot (at 24bit) ?
Just curious.

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Not really.

That DAW fader is way after your audio has been converted to digital, you want to lower the level hitting the converter front end (ie your interface), typically this level can be changed with a rotary knob on the uit or digitally via the unit's control software, either way before you hit the DAW.

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[quote name='51m0n' timestamp='1392382495' post='2367967']
Not really.

That DAW fader is way after your audio has been converted to digital, you want to lower the level hitting the converter front end (ie your interface), typically this level can be changed with a rotary knob on the uit or digitally via the unit's control software, either way before you hit the DAW.
[/quote]

Yep, I understand that, and that was not what I was asking.
Where can the input trim on the DAW channel or buss (or even a trim vst) come into play?
Is it just lowering the volume of the recorded Audio file before going into the channel ?
Or have I just answered my own question ? :)[size=4] [/size]

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Sorry, I was not being clear.
This is assuming you have not gone over with the recording, and the file is good,
but [size=4]quite hot, say -3/-6 or whatever.[/size]
[size=4]It seems trim on the DAW or as a plug in can be useful for gain staging (one way or another).[/size]

Sorry for the thread jack folks, carry on... :)

[size=4]I found an article here about it.[/size]
[size=4][url="http://www.harmonycentral.com/t5/Recording-Video/Recording-Levels-And-Gain-Staging/ba-p/34646453"]http://www.harmonyce...g/ba-p/34646453[/url][/size]

[size=4]And a video.[/size]
[url="http://therecordingrevolution.com/2013/05/01/gain-staging-your-daw-with-trim-plugins-video/"]http://therecordingr...-plugins-video/[/url]

Edited by lowdown
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No need to apologies, this is all helping. :)

That video is good. It was an eye-opener for me to see just how low the volumes were on all of the tracks. I've been trying to get mine all as high as I can without clipping. Then when he showed the master track at the beginning clipping badly I thought, that's just like most of mine. And what have I been doing? I've just chucked a limiter on there and... well yuk.

A question though. Any idea why he's using a trim plugin and not just pulling the faders down? Or do they not do the same thing?

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[quote name='Mornats' timestamp='1392420169' post='2368711']...A question though. Any idea why he's using a trim plugin and not just pulling the faders down? Or do they not do the same thing?[/quote]

Two reasons, really...

1 - the 'trim' applies to the signal at the very entry into the console, and adjusts, or optimises, the level right at the start. Further in, there'll be plug-ins, which will give their best only if this initial signal is correctly set. The fader is too late, and will only be of benefit to the feed to the Master bus.

2 -The best sensitivity of the faders is (usually...) at the top end of their travel. A slight adjustment there is finer than near the bottom. The highter up (within reason...) the fader, the more 'fine control' is available. If one is compensating for a too hot entry signal with the fader by bringing down, one is losing out on this extra sensitivity.

There's doubtless more to it even than this, but that's all I can offer. Subject to completion, correction and/or contradiction from others; hope this helps.

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