Have $1000 of decent gear I don't know how to use properly -- Please help

[DNAInstant]

I'm seeking help on how to use the gear that I have had sitting around for years effectively to actually make music. Here is a list of my gear:

• 24 Fret PRS SE Guitar (basic Humbucking guitar, I think the pickups sound nice)
• Fender Superchamp X2 (Tube pre-amp to digital modeling)
• 2 Shure SM57s. (Been told to only use one to start out with before incorporating anything else).
• Mackie Onyx Blackjack (Pre-Amp with two 1/4"/XLR input/outputs, headphone input)
• Modern Windows 7 PC with Reaper, Audacity, EZDrummer, and many amp simulator programs.
• Been playing guitar (a lot) for 7 years.

Here's my current workflow:

• Plug in all equipment, dial in tone on amp. Then roll back distortion 2-3 notches from where I was.
• Position one SM57 about 1 inch away from slightly off center of the cone of my guitar amp.
• Find the amount of gain for the pre-amp to give an input signal which is just below going "into the red" on Reaper. Usually this is around 35-40 decibels(!?).
• Create click-track in Reaper. Practice tune/riff/whatever a few times, work out what I want to record.
• Attempt an actual take as cleanly as possible.
• Go back and listen to the take -- if sloppy or inaccurate, try again. (<--- one thing I would like to do here is simply play multiple times on repeat until I get a take that I think sounded good, since this is a lot more efficient than having to hit stop/delete/record every time, but I don't know how to do this).
• Usually takes me about 30 minutes to get to this point. Which is a lot, which shows I am a newbie.

As you can see, I've thought about this and researched quite a bit over the years. I will post some sound clips as soon as possible. I am getting way too much mush in my tone and none of the guitar notes are distinguishable from one another. I've tried rolling back the gain on my amps, moving around the mic, etc, but none of this seems to help. The sounds I am getting from my amp are simply not being translated effectively at all into recording, and simply layering even just 2 guitar tracks would I think result in an incomprehensible mess.

Now, I would like to shoot for a simple underground metal sound similar to this: https://www.youtube.com/watch?v=WbRhnCBHab0. Absolutely nothing fancy going on there, production wise (shouldn't be hard to do). One caveat is that I have to record at lower volumes, can't be blasting anything too loud, I would like to keep the amp at around a 2 maximum. Realistically, do I need to build an isolation cabinet to do what I want to do? I honestly don't know, but it is a project I am willing to undertake. Basically it's time to start taking my music a little bit more seriously, frankly.

 

[Bobbsy]

There's no single magic recipe for getting the sound you want--it's a matter of experimentation.

You need to play with the balance between the amp level/sound (distortion etc.) and the mic placement. First off, listen critically to the actual sound you're hearing direct from your guitar amp, with your ear as close to the speaker as possible so you're not hearing it with the added effect of the room tone. Quite seriously, you may want good professional ear plugs (the kind that lower level without affecting the frequency response) to avoid hearing damage!

Once you're happy with the sound coming from your amp, play with the position of the microphone. Mic position on a guitar cabinet is critical and even small moves can make quite a big difference to the sound. I might try moving the mic a bit farther out from the cabinet--I'd definitely try moving the mic a bit farther back--say between six inch and a foot and maybe a bit more off centre with the mic angled to be pointing more towards the centre (my theory being that, as close as you are, you're only getting part of the speaker cone.

The main thing though is to experiment and keep experimenting until you hear something you're happy with. Guides and online advice can only go so far and you get to the stage where you have to use your ears!

 

[ecc83]

"an input signal which is just below going "into the red" on Reaper."

What!!! You should not be anywhere NEAR "red" on the Reaper meters!

You should be recording at a sample rate of 44.1kHz and at 24bits (it hardly matters I know for noisy fekkers like guitars but get into the habit) .

The average level to aim for is -20dBFS on the DAW meters (some folks say -18dBFS but that really only applies for matching "stooodio" equipment levels and neg 20 is good enough for jazz).

With regard to stopping and starting. I don't use Reaper but I bet it works like Samplitude? Record then hit the space bar to stop. Recover composure, start again and Reaper should update the recording with "take 2" and so on. You have an "infinite time tape recorder" just let it roll and then dispose of the chaff later.

Shoot! No, Reaper doesn't work like that*. Ok, so just leave things recording, stop, shout "take N!" start again. You will still be able to go back and find the money take.

*But I bet it can be made to?

Dave.

 

[RAMI]

"an input signal which is just below going "into the red" on Reaper."

What!!! You should not be anywhere NEAR "red" on the Reaper meters!

You should be recording at a sample rate of 44.1kHz and at 24bits (it hardly matters I know for noisy fekkers like guitars but get into the habit) .

The average level to aim for is -20dBFS on the DAW meters

Exactly. Pretty much everything else you're doing shows that you did some reading, which is a good thing. But you don't want your levels to go ANYWHERE NEAR "red" in digital recording. Or, actually, let me re-phrase that since there are people that will say there's nothing wrong with going near clipping as long as you don't actually clip. So......there's no good reason to get anywhere near clipping. Keep your levels conservative and you're not inviting problems.

 

[DNAInstant]

Exactly. Pretty much everything else you're doing shows that you did some reading, which is a good thing. But you don't want your levels to go ANYWHERE NEAR "red" in digital recording. Or, actually, let me re-phrase that since there are people that will say there's nothing wrong with going near clipping as long as you don't actually clip. So......there's no good reason to get anywhere near clipping. Keep your levels conservative and you're not inviting problems.

Okay, I'm getting a lot of good info here. I've tried not going close to the red as mentioned (just by chance/experimentation), but then the track that I get is very very faint and not audible -- in other words, if I were to export it to an mp3, I would have to turn the volume on my headphones all the way up to hear it. Any idea why that would be? In other words, if my input signal is not huge, then my track ends up very small. And that's also why I mentioned perhaps building an isolation cabinet, so my amp volume can simply be louder.

 

[RAMI]

Okay, I'm getting a lot of good info here. I've tried not going close to the red as mentioned, but then the track that I get is very very faint and not audible -- in other words, if I were to export it to an mp3, I would have to turn the volume on my headphones all the way up to hear it.

That's totally normal and we all deal with the same thing. During the tracking and mixing stages, we all have to turn up our monitors A LOT compared to when we're listening to finished, commercial CD's, Youtube videos, or anything else that has already gone through the mastering phase.

Don't worry about volume while you're recording and mixing. You can get your volume during the mastering stage. I'm not saying that volume is the only purpose of mastering, but mastering is the stage where you'll put a limiter on your final mix and get the volume up.

 

[DNAInstant]

That's totally normal and we all deal with the same thing. During the tracking and mixing stages, we all have to turn up our monitors A LOT compared to when we're listening to finished, commercial CD's, Youtube videos, or anything else that has already gone through the mastering phase.

Don't worry about volume while you're recording and mixing. You can get your volume during the mastering stage. I'm not saying that volume is the only purpose of mastering, but mastering is the stage where you'll put a limiter on your final mix and get the volume up.

I've done some basic mastering in Adobe Audition 1.5, but that was quite some time ago. I'll have to look up some tutorials on how to do mastering in Reaper. I didn't find it to raise the volume much in my process, but then again I don't know what a limiter is so perhaps I was not doing the right things.

 

[ecc83]

Now see, I read this a lot!

That people's recordings when done at the correct level of -20ish dBFS play back far too quietly and I just don't get it?

I am presently listening to live Radio 3. Elgar, just piano and violin. On my 2496 card meters the signal is very roughly averaging neg 20 but, being live "classical" music, levels are all over shop but max peak is -10dB. Applause peaks to -15 and the continuity voice back to -20.

Holding aloft a sound level meter (£15 cheapo) I read very roughly 75dBC SPL at my seat. My monitors have a mark to produce 85dBC SPL for -18dBFS but they are almost always backed off from that for "domestic" reasons. The piano and violin sounded at about the right level to me in a very small room.

Now, had the signal been peaking close to 0dBFS it does not take a maths genius to work out that the SPL would be 95dB and well over 100dB if the monitors were set to their calibration marks! Those sound levels are totally beyond what you would get from those two instruments at a good seat in a concert hall and in fact 100dB has long been the max SPL we experience from a full orchestra at all the FFFs!

So, you start at the OUTPUT. Pink noise into monitors to get 85dBC at the listening point and that level should be being reproduced when the DAW is running tracks at -18dBFSish!

Mr Massive Mastering has a link to the whole process.

Dave.

 

[RAMI]

That people's recordings when done at the correct level of -20ish dBFS play back far too quietly and I just don't get it?

I'm pretty sure that if we asked pretty much everyone here that records and mixes at proper levels, everyone would say that they have to turn up their monitors compared to a commercial CD or "finished" product. Maybe not if we're comparing rock to classical, but it's pretty much an accepted fact (?) that, before mastering, everything's pretty low (relative term, I realize). Anyway, speaking just for myself, that's the case.

I admit that I didn't understand most of your experiment above, so I might be missing something.

 

[mjbphotos]

[*] Go back and listen to the take -- if sloppy or inaccurate, try again. (<--- one thing I would like to do here is simply play multiple times on repeat until I get a take that I think sounded good, since this is a lot more efficient than having to hit stop/delete/record every time, but I don't know how to do this).
[*] Usually takes me about 30 minutes to get to this point. Which is a lot, which shows I am a newbie.

In Reaper if you keep the track armed (for recording), and hit stop (and it asks if you if you want to save, click on 'save all') then page back to the start, a new TAKE will automatically be created in the same track when you hit the record button again. Then learn how to blend TAKES, or how to manipulate them.

 

[ecc83]

I'm pretty sure that if we asked pretty much everyone here that records and mixes at proper levels, everyone would say that they have to turn up their monitors compared to a commercial CD or "finished" product. Maybe not if we're comparing rock to classical, but it's pretty much an accepted fact (?) that, before mastering, everything's pretty low (relative term, I realize). Anyway, speaking just for myself, that's the case.

I admit that I didn't understand most of your experiment above, so I might be missing something.

It wasn't an experiment Rami, it was an example of how my system is setup and I cannot conceive of any other way to do that!

But perhaps things would be plainer if I start from first principles ?(wtgr)

My Tannoy monitors are said to have a max SPL at one mtr of 109dB. Let us take that as a wee bit optimistic and call it 100dB?

Now I know that the max output of my soundcard occurs when the software hits 0dBFS. So, it makes abundant sense to have the two figures roughly aligned?

We now have our ultimate output and max headroom sorted and convention has always used 20dB or thereabouts as headroom ( mixers OP level +4dBu max out +22-+26dBu) and that brings us back to average, meat and spuds, level of -20dBFS which equates of course to about 80dBSPL listening level.

The precise figures are of course on record and have been chosen for psychoacoustic reasons and "fatigue" issues. The home recording noob will find 83 or 85dB too loud for monitoring in many social settings but that does not matter. You can turn down the monitors to 70dB or so. What matters is that you HAVE calibration marks and can go to it for critical listening.

Lastly, all this does not come "strange" to me because I trod the "hi fi" path when much younger (until it went all Russ A and silly) and recording for me was capturing the best "reality" I could. If I record an acoustic guitar I fekking KNOW how loud it was in the room and I expect it to reproduce at that level on play back.
The snag comes of course when you are recording forces beyond those that could exist in a small room and/or are beyond the capabilities of the reproducing equipment. This is where the skill of al involved in such recordings comes in. Fooling us (a bit!) that we have the Liverpool phil' or Prodigy just beyond those feeble speakers!

Dave.

 

[RAMI]

I can't argue with any of that, and I'm not arguing with it either way. One of the reasons is because I don't understand it. That's not your fault, it's mine. I'm sure everything you're saying makes sense. But I don't understand SPL levels, dbu's, dbsfls, dbfsm's, dbspl's, or any of that. Also, I have the attention span of a 4 year old. I'm reading your post, trying my hardest to follow it, and by the third paragraph, my eyes are getting fuzzy and I'm reaching for my little toy fire engine.

All I know is that when I'm listening to any commercial CD, video on Youtube, or MP3, the volume knob on my receiver is at between 9 o'clock and 10 o'clock. When I'm recording and mixing my music, I have to turn the volume knob up to between 12 o'clock and 1 o'clock. Maybe someone else can explain what I'm saying and also explain the reasons why. But there is no way anyone's going to tell me that an un-finished project is as loud as a finished one. That's all I know.

 

[JLondon]

Sometimes I take an unfinished project and just export it to an mp3 or wave file and "normalize" the audio upon exporting which brings up the volume all at once adjusting for peaks - just so I can listen to it temporarily. This adds about 30db to the exported file!

 

[ecc83]

Ok, so I am playing a "proper" CD. Miles Davies' Greatest Hits.
Sure, it has all been brought up to hit a gnat's undercarriage below 0dBFS but that was for track 1, Seven Steps to Heaven. Track 3, Someday my Prince Will Come is more reflective and peaks (pk hold set in Samplitude) never go above -3.
Track 4, Walkin' crowds zero again at the start but parts are "allowed" to run way down at -20!

Naturally I had to pull the volume down feeding my monitors to keep this music at tolerable (C 11pm) levels.

When mixing, if every track was at the same level they would add about 3dB each to the mix so there has to come a point where the total mix has to be reduced in level. Note, it is virtually impossible to overload a "DAW" internally but you can overload the output of the soundcard and, I understand certain plugins?

Of course, for amplified rock and electronic music there is no "real life" reference.

Dave.

 

[Bristol Posse]

-20 dBFS is just a meaningless arbitrary number without knowing how converters are calibrated. It will give you completely different voltages at the analog/speaker outs depending on how your converters are calibrated.

If you have high headroom converters then sure -20 dBFS could be around line level 0VU (+4dBu) and if you are using monitors that run at line level then sure the levels would sound OK. If you are running a cheap interface or the line out of your computer sound card which typically have pretty low DA headroom then -20dBFS could be 16dB below 0VU and would be barely audible through the monitors

Headroom is a good thing if your equipment is built with headroom. If it's not built with headroom but you still need to get to around line level to get the best of signal to noise ration (ie a lot of cheap interfaces) then you may need to record hot and then pull the faders down in the DAW to get the mixing headroom back. I don't see any highly specced converters in the OP

line level = -18dBFS is only true if you have converters calibrated to max output (0 dBFS) = +22dBu

Many cheap/entry level converters max out at +12 dBU or even less (some even below +10dBu) in which case you have a lot less recording headroom to get optimal S/N ratio and you need hotter levels in the DAW or a lot more gain in a monitor amp to get audible playback

on size does not fit all unfortunately

 

[ecc83]

Attached is pink noise at -20dBFS and it reads -20 as the original.wav file generated in Adobe Audition 1.5.

In the absence of a sound level meter it should play quite loudly being louder than you would have the telly for a soap. It should be intrusive enough to be slightly difficult to talk over.

a monitor system should, in my view, reproduce -20dBFS recordings at this loudness level as a "standard" but, as I have said before, we can't always do that in practice and of course commercial tracks that have been squashed to buggery will come out very loud indeed!

But! Sanity is it seems prevailing. The industry is going over to "loudness" measurements and under this regime very low dynamic range recordings, running at the limit will, it seems sound very "flat" and unexciting.

I might, in the future, be able to understand film dialogue!

Dave.

 

[ecc83]

-20 dBFS is just a meaningless arbitrary number without knowing how converters are calibrated. It will give you completely different voltages at the analog/speaker outs depending on how your converters are calibrated.

If you have high headroom converters then sure -20 dBFS could be around line level 0VU (+4dBu) and if you are using monitors that run at line level then sure the levels would sound OK. If you are running a cheap interface or the line out of your computer sound card which typically have pretty low DA headroom then -20dBFS could be 16dB below 0VU and would be barely audible through the monitors

Headroom is a good thing if your equipment is built with headroom. If it's not built with headroom but you still need to get to around line level to get the best of signal to noise ration (ie a lot of cheap interfaces) then you may need to record hot and then pull the faders down in the DAW to get the mixing headroom back. I don't see any highly specced converters in the OP

line level = -18dBFS is only true if you have converters calibrated to max output (0 dBFS) = +22dBu

Many cheap/entry level converters max out at +12 dBU or even less (some even below +10dBu) in which case you have a lot less recording headroom to get optimal S/N ratio and you need hotter levels in the DAW or a lot more gain in a monitor amp to get audible playback

on size does not fit all unfortunately

Does not matter in this context.

I am not talking about converters or their calibration levels.
I have just put noise at 0dBFS thru my 2496 and it produces about 0.43V out that's about -7dBV or -5dBu. That actually feeds a small mixer that then feeds my Tannoy 5A monitors. So long as there is enough gain in the system (and there is!) I can have my mixer hitting -20VU for -20dBFS and my monitors kicking out 85dBSPL . It is far from a "pro" mixer but it can certainly manage a clean +10VU and 95dBSPL will do for me!

Of course top line converters run at pro levels and there would be a corresponding lowering of the gain in the monitor chain for 85ish SPL . In fact many active monitors have too much gain even for home recording bods which is why some are rather noisy!

Dave.

 

[Bristol Posse]

Does not matter in this context..

You said you couldn't understand how anyone could possibly not find -20dBFS loud enough. I explained why it my be barely audible in some systems

It' doesn't matter to you on your setup, but the OP is not running your set up and being able to hear his recording matters to him. If he has a low head room DA then running Levels at -20 dBFS may not work for him

 

[ecc83]

You said you couldn't understand how anyone could possibly not find -20dBFS loud enough. I explained why it my be barely audible in some systems

It' doesn't matter to you on your setup, but the OP is not running your set up and being able to hear his recording matters to him. If he has a low head room DA then running Levels at -20 dBFS may not work for him

No! The OP is running a setup with considerably more output than mine!

The Mackie Blackjack has a max out of +10dBu (~2.5V rms) a good 15dB above my sound card. He does not mention his monitors AFAICS but there can surely be none so insensitive that a couple of volts won't hit their maximum? My 5As need 0dBu for full output and they are on the low side for sensitivity, many monitors will max out for neg 10dBV.
The Mackie has a quoted dynamic range of 109dB, 10dB better than my old 2496 but DR does not matter here. Everything in a "studio" should be set up such that the feed to the monitors is at a standard level be it a 2496, a Prism or a 60dB DR tape machine!

I have had a variety of interfaces through my hands in the last 7 years and have never had a problem getting sufficient volume from my monitors from recordings done at -20dBFS. I can even drive them perfectly well from the OBS jack on my HP i3 laptop.

You could drive the 20,000W (say) sound system at Glasto' to insane levels with an iPod, all you need all you need is the gain.

If a system cannot produce clean 80dBSPL levels for -20dB recordings there is something awry in the monitoring gain set up.

Dave.

 

[Bobbsy]

To try and drag this kicking and screaming back to the original question...

While the original poster is probably tracking too hot, as long as he's not clipping the signal this is almost certainly NOT the cause of the "mush" and lack of detail he's talking about. Tracking at an appropriate level leaves room for later mixing with other tracks and the use of plug ins that may or may not have problems with high levels. The mush is much more likely to be something to do with mic placement and/or the acoustics of his recording space.

Now, onto my thoughts about the discussion this evolved into.

Basically it needs to be remembered that there are several different "levels" in the production process.

-You TRACK at a level that leaves sufficient headroom for later mixing and plug ins. Some say -20, some say -18, I tend to worry more about the occasional peaks which I never want higher than -8 or so with the "average" falling where it does. There's no right or wrong. None of my plug ins seem to worry about levels approaching 0dB(FS) and, especially working in Floating Point, I can just move a fader up and down if adding all the tracks together seems to get too hot.

-You MONITOR at a level that's comfortable to listen to and appropriate fotr the musical genre. Yeah, there are set down suggestions but the main thing is that you can turn your monitors up and down without affecting the actual levels in the mix. Turning up your monitors doesn't affect the actual levels any more than turning up your TV changes things at the TV station.

-You MIX so that you still have some headroom left for any processes that will be used in the mastering process.

-You MASTER to (amongst many other things) make the final levels on your project something similar to a commercial recording. Almost always this'll mean peaks right at the 0dB(FS) but you can leave more or less dynamic range (i.e. the difference between the loudest and quietest parts of the mix) down to your taste. Too little dynamic range can make your music sound flat and just generalised noise. Too much and the quiet bits will sometimes get lost in today's noisy listening environments.

Somebody mentioned making CDs of a work in progress and normalising to get levels up to something listenable in a car or whatever. Indeed, this is exactly what I do when I want to listen to something to check the mix before doing the whole mastering routine.

No, you don't want to clip. Yes, you want your mixes loud enough to hear in a car or on earbuds on the train--but, between these points, there's a lot of wiggle room and no big rights or wrongs.