Recording music is driving me nuts!!! HELP!

Pulling out your hair? Don't know what to do or where to go? Ask in here.
Forum rules
READ: VSE Board-Wide Rules and Guidelines

If your Help request has been solved, please edit your first post in order to select the Image Topic Icon to let others know your topic has been solved.

Recording music is driving me nuts!!! HELP!

Postby JSRockit » Tue Nov 14, 2006 4:17 pm

First question. I record all of my seperate synths and drum machines at their optimum levels, then I mix them in logic... however, to make them sound loud and powerful...I have to shove everything into the red...even with compressors and EQ. That's fine, but when it comes to limiting the stereo mix out...it never sounds right... everything sounds dead...crushed, lifeless. My questions is, should I just stick with my low levels when mixing, make sure nothing goes into the red (no matter how anemic the mix sounds), then at the end try to gain volume with a limiter? I'm lost here. What is the proper way?

Also, I am not talking low vs. modern radio songs... I'm talking way lower than anything should be... I'm talking turn the volume way up on my computer speakers (my second source for listening vs. Dynaudio BM5a's) and it still isn't loud enough.

Lastly, I noticed that with the BM5a monitors, I have to have the bass as almost inaudible for it to sound right on other systems...systems with subwoofers. I have A/B'd it with other professional CDs and the bass is the same (almost inaudible) on those CDs as well. Why is this?

I'm really to the point where this is more stress than its worth. I love making music, like writing lyrics, but I can't stand recording...it takes all the fun out of everything. However, without recording...I have no reason to be making my songs.
I have read numerous tutorials and stuff that others have pointed me towards...but nothing seems to get it through my head. I really have no excuses equipment wise... people make great mixes and great recordings with so much less. I'm using Logic Express, Izotope Ozone,
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby konvert » Tue Nov 14, 2006 4:39 pm

Wait... I'm not sure if I understood this one part of your story: Are you _recording_ your synths/stuff WITH compressor and EQ on??
Because if you are, stop that. I might've misunderstood you. Actually, now that I reread it, I'm guessing you are not, so that's fine.

That sounds little funny however, that you have to turn everything up so much.
What do you exactly mean by 'recording at optimal level' ? Recording as loud as you can without clipping is the way I have learnt to do it at least. But nowadays I don't even do that, and I seem to not have problem with overall level.

Could you put some little snippet of your track up somewhere, _without_ limiting?
It seems that you are limiting too much, if your track sounds squashed and lifeless.
Vermona DRM1 mk2, Casio CZ-1000, MaM VF-11, Boss PC-2, Korg Minipops Junior
User avatar
konvert
Junior Member
Junior Member
 
Posts: 156
Joined: Fri Sep 16, 2005 10:38 am
Location: Finland

Postby JSRockit » Tue Nov 14, 2006 4:44 pm

konvert wrote:Wait... I'm not sure if I understood this one part of your story: Are you _recording_ your synths/stuff WITH compressor and EQ on??
Because if you are, stop that. I might've misunderstood you. Actually, now that I reread it, I'm guessing you are not, so that's fine.

That sounds little funny however, that you have to turn everything up so much.
What do you exactly mean by 'recording at optimal level' ? Recording as loud as you can without clipping is the way I have learnt to do it at least. But nowadays I don't even do that, and I seem to not have problem with overall level.

Could you put some little snippet of your track up somewhere, _without_ limiting?
It seems that you are limiting too much, if your track sounds squashed and lifeless.


I'm recording as loud as I can without clipping, with no effects during the recording stage. Then I mix, then I add compressors and EQ, and at that time... I notice that my master level is in the red. So, then I try to get it out of the red by using a limiter...but that is just squashing the h**l out of it. I guess I should bring the master fader down so it isn't in the red, then try a limiter to gain volume?
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby konvert » Tue Nov 14, 2006 4:54 pm

JSRockit wrote:I'm recording as loud as I can without clipping, with no effects during the recording stage. Then I mix, then I add compressors and EQ, and at that time... I notice that my master level is in the red. So, then I try to get it out of the red by using a limiter...but that is just squashing the h**l out of it. I guess I should bring the master fader down so it isn't in the red, then try a limiter to gain volume?


Ok. You are fcking up the mixing, then. Use compressors and eq's sparingly; not every channel needs them. Actually that's the first stage which will lead to your sound being lifeless. For example, if you're using some ethereal, airy pads (just an example), you reall don't need to compress them, otherwise they, well, lose their airiness and all life.

Ok, so let's say you compress your drums and bass-line then.
You need to bring down the individual track-levels, not your master if your master starts to go on red.
Also make sure to EQ your tracks right; you should use cut way way more than boost. For example, you can cut pretty much bottom out of your vocals so they will not mess your bass and drums.

Low frequencies eat much of the overall power of your song.
Actually I almost always cut everything away below 30hz, you pretty much can't hear it anyway but if they are left there,
you'll find trouble at the final stage, i.e. 'mastering'

Uhh, gotta run now but i'll be back soon.
Vermona DRM1 mk2, Casio CZ-1000, MaM VF-11, Boss PC-2, Korg Minipops Junior
User avatar
konvert
Junior Member
Junior Member
 
Posts: 156
Joined: Fri Sep 16, 2005 10:38 am
Location: Finland

Postby JSRockit » Tue Nov 14, 2006 5:19 pm

konvert wrote:Ok. You are fcking up the mixing, then. Use compressors and eq's sparingly; not every channel needs them. Actually that's the first stage which will lead to your sound being lifeless. For example, if you're using some ethereal, airy pads (just an example), you reall don't need to compress them, otherwise they, well, lose their airiness and all life.

Ok, so let's say you compress your drums and bass-line then.
You need to bring down the individual track-levels, not your master if your master starts to go on red.
Also make sure to EQ your tracks right; you should use cut way way more than boost. For example, you can cut pretty much bottom out of your vocals so they will not mess your bass and drums.

Low frequencies eat much of the overall power of your song.
Actually I almost always cut everything away below 30hz, you pretty much can't hear it anyway but if they are left there,
you'll find trouble at the final stage, i.e. 'mastering'

Uhh, gotta run now but i'll be back soon.


Ok, I do most of that...but tend to make my bass drums and snares go into the red...to make them sound big...because I cannot get them that way any other way. So I guess I need to bring them down to just below red, then mix the rest (synths, samples, melodies) based on that. My pads and strings have no compression. I do cut out the low-end frequencies on everything...including vocals with a high pass filter on each track.
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby code green » Tue Nov 14, 2006 6:00 pm

you could probably make a lot of cuts on the drums and bass, so you get more out of the "good parts" of the eq spectrum that you want from them. (i.e., so you don't have to cut the overall levels as much on these tracks).

set up a notch eq with a boost of about +10...then sweep it slowly across the spectrum on a soloed track. note where it sounds not so good and then go back and make small cuts in these places. where it sounds good, give 'er a boost.
User avatar
code green
Active Member
Active Member
 
Posts: 535
Joined: Wed Mar 01, 2006 6:58 am
Location: brooklyn
Gear: prophet 600/evolver/juno 6/alpha juno 2/bassst'n/crumar performer/jv1010/suitcase rhodes 73/ '78 gibson l6-s/'73 guild mahogany/'69 fender princeton
Band: thermite zapruder

Postby JSRockit » Tue Nov 14, 2006 6:14 pm

code green wrote:
set up a notch eq with a boost of about +10...then sweep it slowly across the spectrum on a soloed track. note where it sounds not so good and then go back and make small cuts in these places. where it sounds good, give 'er a boost.


I have no idea how to do this...nor do I have any idea what frequencies to cut.
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby konvert » Tue Nov 14, 2006 6:23 pm

JSRockit wrote:
code green wrote:
set up a notch eq with a boost of about +10...then sweep it slowly across the spectrum on a soloed track. note where it sounds not so good and then go back and make small cuts in these places. where it sounds good, give 'er a boost.


I have no idea how to do this...nor do I have any idea what frequencies to cut.


You can use graphical eq for that; Increase the Q-value to the max, then boost all up so you see very narrow spike. Then start to sweep the frequencies and when you find a spot which sounds really bad (when boosted, that is), like ringing or booming, then you just cut (decrease the gain) on that point.

What equalizer(s) do you use then? Logic's own?

http://magnus.smartelectronix.com/#effects

and find Nyquist EQ. It's parametric graphic equalizer.
Vermona DRM1 mk2, Casio CZ-1000, MaM VF-11, Boss PC-2, Korg Minipops Junior
User avatar
konvert
Junior Member
Junior Member
 
Posts: 156
Joined: Fri Sep 16, 2005 10:38 am
Location: Finland

Postby Altitude » Tue Nov 14, 2006 6:36 pm

I'm recording as loud as I can without clipping


That's your problem, your recording WAY too hot so you have no headroom left after recording. Modern A to Ds are designed to be used at about -18 dBFS which is not even close to what your doing. try recording -18 to -12 dBFS and see what happens
User avatar
Altitude
Expert Member
Expert Member
 
Posts: 1220
Joined: Sun Sep 24, 2006 7:25 pm
Location: Michigan

Postby JSRockit » Tue Nov 14, 2006 6:58 pm

Altitude wrote:
I'm recording as loud as I can without clipping


That's your problem, your recording WAY too hot so you have no headroom left after recording. Modern A to Ds are designed to be used at about -18 dBFS which is not even close to what your doing. try recording -18 to -12 dBFS and see what happens


Really? If I do that, won't my mixes even be alot lower in volume?
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby JSRockit » Tue Nov 14, 2006 6:59 pm

konvert wrote:
What equalizer(s) do you use then? Logic's own?

http://magnus.smartelectronix.com/#effects

and find Nyquist EQ. It's parametric graphic equalizer.


I use the Logic ones and the Ozone one.
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby konvert » Tue Nov 14, 2006 7:02 pm

Altitude wrote:
I'm recording as loud as I can without clipping


That's your problem, your recording WAY too hot so you have no headroom left after recording. Modern A to Ds are designed to be used at about -18 dBFS which is not even close to what your doing. try recording -18 to -12 dBFS and see what happens


I don't agree with that. You can always lower the volume on tracks after you've recorded them.

To JS:

The point is not to have each track to play as loud as possible, the point is to find a balance between them.
Vermona DRM1 mk2, Casio CZ-1000, MaM VF-11, Boss PC-2, Korg Minipops Junior
User avatar
konvert
Junior Member
Junior Member
 
Posts: 156
Joined: Fri Sep 16, 2005 10:38 am
Location: Finland

Postby JSRockit » Tue Nov 14, 2006 7:08 pm

konvert wrote:To JS:

The point is not to have each track to play as loud as possible, the point is to find a balance between them.


Ok, that's my homework for tonight...to go home, reduce the levels on all the individual tracks (so none are in the red at all), mix them to cooperate with each other ie so they sound balanced, and overall, until the master level is also no longer in the red. Does that sound like a good place to start?
Korg Volcas / 6 x TE POs / MicroBrute / EH Space Drum & Crash Pad
User avatar
JSRockit
Synth Explorer
Synth Explorer
 
Posts: 3911
Joined: Thu Jan 19, 2006 11:15 pm
Location: New York City

Postby konvert » Tue Nov 14, 2006 7:35 pm

That's a good place to start, yeah.

Concentrate on getting your mix good and only after that start worrying about the overall level. :)
It's frustrating at times, to try to find a good balance and sometimes trying to find the one thing which spoils the track so you don't get it to sound good.
Vermona DRM1 mk2, Casio CZ-1000, MaM VF-11, Boss PC-2, Korg Minipops Junior
User avatar
konvert
Junior Member
Junior Member
 
Posts: 156
Joined: Fri Sep 16, 2005 10:38 am
Location: Finland

Postby Altitude » Tue Nov 14, 2006 7:35 pm

I don't agree with that. You can always lower the volume on tracks after you've recorded them.


Sorry, but you're incorrect. This is one of the biggest and most common mistakes ppl make with digital recordings. Here is a lengthy explanation (not mine):


-- -- -----------------------------
if you record with a 24 bit word, the noise floor is so low that setting levels that peak well below full scale is fine, still way above the noise floor.

Each bit you add to the word doubles the available values the word can represent, and therefore doubles the dynamic range (signal to noise ratio from full scale down to noise) that you can record.

A doubling of dynamic range equates to 6db. Therefore, each bit in the word contributes 6db of dynamic range. A 16 bit word therefore has a 96db signal to noise ratio, and 24 bit word can express 144db of signal.

In the real world, the audio electronics in the converter provide a higher noise floor than a 24 bit word can represnt, so a good 24 bit converter will give, lets say conservatively, 110db of signal to noise.

This means that if you record your audio with peaks no higher than -14db under Full Scale, you'll still be experiencing a recording with 96db of dynamic range, which is the best any 16 bit CD has every accomplished.


To make the point even more graphically - this all assumed that the source signal has a dynamic range in excess of 96db too. I would bet you a beer that it isn't even close. There's no tube mic that operates that cleanly. Your studio room has noise higher than that. All your hardware compressors and EQs operate with a much higher noise floor.

If you were very careful, and ended up having a source with 70db of dynamic range (congratulations!) you could record it with peaks at -26dbFS (-26 under full scale) and still have preserved every ounce of dynamic range.

So its obvious that hitting full scale isn't necessary at all - why not preserve some headroom just in case? Let's say you do make it just under full scale. No harm in doing that if you don't go over, right?

Well, what do you do when you want to EQ something +2db? Where does that 2db go? Into clipping of course, unless you lower the input level of the plug in, which is going to lose any hypothetical S/N benefit you had preserved anyway.

Even more importantly, when you record this hot, I've got to ask - what did you do to your preamps, and analog chain to get this level? Most converters are set so that 0dbVU = -18dbFS.

That means that if you're getting -6 below full scale on your converter, that you're +12db over the 0VU point! Many analog electronics can c**p out here, but almost all will sound different at least. Some times it may be "better" but usually, its a small accumulation of distortion that builds into a waxy fog that makes people blame "digital recording" for its pristine playback of their slightly distorted, but "pretty on the meter," tracks.

If you record with levels around your 0 point, some thing like -18dbFS or -14dbFS, depending on how your converter is calibrated, you'll have your analog electronics in their sweet spot, headroom for plug ins and summing, an appropriate analog friendly level if you use analog inserts later in the process, and on a modern 24bit converter with 110db S/N, the ability to faithfully record signals with a dynamic range of over 90db.

And by all means, 0dbVu is no glass ceiling like 0dbFS is. Keeping levels around 0dbVU doesn't mean that peaks won't exceed that by 6 (or more) db. If they do, your ability to record 96db of S/N (if you even have it in the source, and you don't), just like the best CD you ever heard will be preserved, if peaks don't exceed -12dbFS! More if they do.


) A 24 bit PCM word can express a theoretical limit of 144 db of S/N.

2) The analog electronics in the converter limit the performance to a functional 100 db of S/N. (slightly more in some cases, but I'll use a conservative figure and make the point even without those extra 6 db)

3) As long as the noise floor in any recording system is lower than the noise floor in the signal you're recording, you will record the full dynamic range perfectly.

4) No source you've ever recorded had a signal to noise ratio higher than 80 db, and most would be much much lower. Lynn suggests that he RARELY sees the source's noise floor lower than 70 db down, and even then, rarely. Assuming that his peaks are not at full scale, his typical source S/N ratio must be in the 50-60 db range?

This means that if you record your (best ever) 80db S/N source into a converter so that the highest peak just reaches -19 dbFS (below full scale) on the meter, that the noise floor in your signal will be louder than the noise floor in the converter. You needn't record it any hotter than that.

In the real world, you could get away with peaks around -28 dbFS, and be PERFECT. Any higher than that is totally unnecessary.

Conclusion: There is absolutely NO benefit to tracking hot.

But does it hurt to do it? Read on...

1) Your microphone preamp is set to perform best (gritty distorted choices aside) peaking around 0dbVU. This is where you'd have it set if you were recording to analog tape, hitting 0 on the VU meter. Plug that same source into most converters, and you get peaks around -20dbFS to -14dbFS, depending on how the converter is setup.

The scientists who developed this system understood the situation, even if the guys who wrote the digidesign manual don't! They EXPECT you to record with peaks around 0VU (-18dbFS on the digital scale). They KNOW about the signal to noise deal I explained earlier. That's why they chose to put the nominal level so "low" on the meter.

When you record hotter, with peaks at -6dbFS, lets say. You're driving your mic preamp 12 db hotter than you did yesterday in the analog world! That's going to add a subtle layer of distortion to your project. And they say analog sounds so much better than digital - maybe its because most people use their analog gear incorrectly when recording to digital. Maybe the "problem with Pro Tools summing" is really the effect of tracking too hot?

I've heard people say "My Neves can handle outputs +24db according to the spec, so what's the big deal?" My Neve 1073s are great sounding workhorses. They are rated for a LOT of gain. Still, they definitely sound very different even at +12. Very different. Maybe a good choice in some cases, but not the norm.

2) If you have a peak at -2dbFS, and you try to boost a mid range frequency +3db on an equalizer, you're going to clip.

Another unintended detriment to tracking hot is that you no longer have any headroom in your plug ins! It is true that in Pro Tools, you can recover lost headroom in the mix bus by lowering the master fader. This isn't true in an analog console, where the distortion has happened in a summing amp "upstream" on the master fader. In that case, the master fader only lowers the volume of the distorted signal, which remains distorted.

In Pro Tools, the master fader is actually a co-efficient with each individual fader before summing. This means that if you're clipping the mix bus, you can pull the master fader down, and fix it. Great. But what about the plug ins across each fader? They aren't affected by the master fader (thank god, or your compression levels would change etc) but neither are they protcted by the master fader. If you're clipping the mix bus, and have your master fader at values lower than unity, then odds are that you're clipping some plug ins too.

3) Most analog gear doesn't like inputs that are 12db and more over 0, even if the spec says they can take it. If you track hot, you're causing a nightmare for analog gear that you may choose to insert during the mix. Keep your levels around 0dbVU, and you can leave the digital domain freely without adding more sonic grunge.

Conclusion: Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

So, to reiterate:

1) There is absolutely NO benefit to tracking hot.

2) Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

If you want to hear the result of tracking too hot, and what it does to Pro Tools, listen to any Lenny Kravitz record. believe me, he uses all the best vintage gear, with gobs of headroom etc. There is no shortage of Neve, Helios, Fairchild, Neumann, Telefunken or whatever on his sessions. The sound of those records is entirely due to the tracking and mixing levels.

"But how do I get my product hot?"

There is a point to having a final mix that peaks at -0.1dbFS. if you are going to have a 16 bit version, if you want to be commercially competitive, if you like to see all the lights light up - sure, I do it every time. The point is i bump it up LAST in plug ins across the master fader. That way, the mix is all properly gain staged, with lots of headroom right up until the last thing juncture. Then if I raise the result to just below clipping after having the benefit of proper levels all the way through, everything is beautiful.

If you are a non believer, try it. The amount of air, detail and image is astonishing. In fact, eventually you may find that Pro Tools is actually TOO CLEAN and transparent! Then you'll start introducing purposeful distortion in your mix - distortion that YOU control at the mix is a very different animal than the unwitting accumulation of crud that comes from tracking too hot all along.

So all this means is that the noise is SO low in a good modern 24 bit converter that you can keep gobs of headroom for proper interfacing with analog gear, and still get the full 96+db of dynamic range, just 12 to 18 db lower on the meter. Your analog gear will thank you too.

So all of this results in a pristine, beatiful, airy, detailed 24 bit mix with peaks around -12dbFS? Cool! .
User avatar
Altitude
Expert Member
Expert Member
 
Posts: 1220
Joined: Sun Sep 24, 2006 7:25 pm
Location: Michigan

Next

Return to HELP!

Who is online

Users browsing this forum: No registered users and 8 guests

cron