Old 12-02-2009, 09:51 PM   #1041
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...Started to look for some info about this and since then, I record my tracks so that they peak between -18dB/-12dB (44.1/24). And you know what? The tracks practically mix themselves...
Without criticizing or endorsing your or anyone else's methods, my point is not that low recording levels are good, nor that high levels are bad, just that gain-staging matters.

All else being equal, recording at -3 or even -0.03 is better: less noise, higher resolution. And with at least one guitar sound that I sometimes use, I actually prefer the sound of a slightly clipped guitar DI feeding a particular amp sim from a particular signal chain.

My point was not that this guy needed a better recipe, my point was that he was following a "recipe" even though it sounded bad, and he was ignoring the fact of the bad sound, because he believed that a little LED was more reliable than his own hearing. All I had to do was point out the obvious artifacts and he (or anyone else) could hear them clear as day.

This is a pretty competent muso/recording guy, and if you asked him in person he'd tell you that I had some special gift or gear for always making everything sound better and more "professional" or whatever, but the fact is that he can hear the differences as well as I can. If he couldn't, he'd never send me any work. But I'm not giving him magic nor cluing him in on any great secret, I'm just pointing out stuff that is obvious to anyone with functional hearing.

It's like that thing in the Bible: they listen, but they do not hear.
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Old 12-03-2009, 03:54 AM   #1042
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Without criticizing or endorsing your or anyone else's methods, my point is not that low recording levels are good, nor that high levels are bad, just that gain-staging matters.

All else being equal, recording at -3 or even -0.03 is better: less noise, higher resolution. And with at least one guitar sound that I sometimes use, I actually prefer the sound of a slightly clipped guitar DI feeding a particular amp sim from a particular signal chain.
Your lesson has been obvious, indeed. What crosses my mind now: is this one of the reasons why some pros tell us how "afwul" digital is sounding? Pilot errors?

With my EMU 1616m and the internal hardware limiter switched on things can get even worse. The DAWs meters will never clip, the EMU's meters will never clip and it still sounds horrible, of course, when the level gets too high.

Sidenote ( I can't resist )

As the highest bit will be used (switched on) whenever the level exceeds more than 50% of the max voltage (-6dB) there is no change in what many of you call "resolution".
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Old 12-03-2009, 04:59 AM   #1043
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Without criticizing or endorsing your or anyone else's methods, my point is not that low recording levels are good, nor that high levels are bad, just that gain-staging matters.

All else being equal, recording at -3 or even -0.03 is better: less noise, higher resolution.
Then now, I'm completely confused. On one hand you wrote this:

"He was flabbergasted to find that the noises disappeared and that the bass sounded vastly better when recorded with peaks at around -12 on the peak meter."

And then, you write the passage I quoted. Wasn't this what your guy was doing, record as hot as possible? I guess I will need to re-visit your gain-staging posts in the zip file because you've totally lost me.

The only sliver of light I see for my own understanding of your last post is this:

Ok dude, so your peak meter never went into the clip zone. Think you're safe? Yeah? Listen again buddy. You hear this ugly thing right there? Yeah? Then bring your recording level down. Does it sounds good, without any ugly artifact? Yeah? Then carry on, you're doing fine.

Am I in the right ballpark here?
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Old 12-03-2009, 05:50 AM   #1044
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What he means is that theoretically, if nothing is clipping or distorting, then recording hotter will get you more resolution.

In practice you should rely on your ears AND the peak meter, because things could be clipping at different points in your signal chain before your converters and you'd never know if you weren't paying attention.

Turning down the last gain stage so the converters don't clip isn't going to save you if you're clipping earlier in the signal chain.
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Old 12-03-2009, 07:13 AM   #1045
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Default Acoustic Guitar...

Ok so semi-related to the clipping thing, but a bit opposite (and I'm an amature so forgive my ignorance). I'm not clipping my recordings, but rather finding it almost IMPOSSIBLE to clip. In other words, I'm getting nowhere near -12 for a recording of my acoustic without really pushing up the gain and making things sound hissy.

I've been trying to record my acoustic through a Firebox and an AT-2020 (yeah yeah, it's a piece of crap but it's what I got). I don't yet have a preamp other than the ones included in the Firebox so I have to deal with it. If I turn up the gain too much I get a lot of hiss from the mic. I can put a gate on it, but that makes the guitar sound a bit too "harsh" for lack of a better word. So I tried recording a stereo track with the acoustic miked on line 1 and a direct input from my pickup to line 2. I then turned the track into a mono track which actually sounded pretty good with a little EQ.

My problem is that usually...unless I really push up the gain (which increases the noise "floor"...the signal comes in at around -24db which seems very low to me. So I've tried doing things like adding chorus to make it a bit more full and then doubling or even tripling the track, panning to left and right, adding reverb, adding an amp plugin, but with each of these I get farther and farther away from the acoustic sound that I want.

Is -24 through a mic normal for an acoustic? Or should it be louder than that?

I realize that a preamp would allow me to increase gain more, but if I have a good recording that's just quiet what kinds of things can I do to make it a bit louder?
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Old 12-03-2009, 09:30 AM   #1046
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Sidenote ( I can't resist )

As the highest bit will be used (switched on) whenever the level exceeds more than 50% of the max voltage (-6dB) there is no change in what many of you call "resolution".
Yeah, I similarly can't resist to get all mathy and shit...

Let's say you record to an average of -12dBFS, which is technically throwing away the top two bits, which is 1/4 of possible information in a 24bit recording. So:

2^24 / 4 = 4194304 possible sample positions

Compare that to 16bit audio, the primary target for recording these days:

2^16 = 65536 possible sample positions

And divide those numbers:

4194304 / 65536 = 64

So, even recording at -12dBFS at 24bits, you are recording at a resolution that is 64 times your ultimate target resolution.

Give yourself some damn headroom, it's okay. And don't record in 16bit...
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Old 12-03-2009, 09:51 AM   #1047
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What crosses my mind now: is this one of the reasons why some pros tell us how "afwul" digital is sounding? Pilot errors?
There's this whole huge EPIC thread (not unlike this one) over at GS that gets pretty deep into this topic:

http://www.gearslutz.com/board/so-mu...log-mixes.html

The gist of it is pretty much what yep says, proper gain staging is hugely important.

There's also a concept, though, that mixing to 0dBFS is just bad practice -- that, in the analog days, you *never* mixed with your VU meters slammed all the way to the right, even though that was the top of your headroom. The "industry standard" for outboard gear was +4dBu/1.23 volts, which is where they set 0VU. If you do the math in the post you come up with a -20dBFS calibration on the digital scale for for 0VU/+4dBu/1.23V. The argument is that most pro quality plugins use -20dBFS their calibration point, so sending your plugins a 0dBFS signal is like sending your outboard gear a 24dBu signal... In other words, a face-meltingly hot signal. Not a problem for digital plugins, mind you, but for analog modeling this can really change the tone of what you get.

This gets pretty deep in the weeds, to be honest. It also goes out the window in the sense that most indy plugin devs don't calibrate to anything in general. I've been experimenting with this concept a bit, though, and frankly I've liked the sound of my mixes with a bit more headroom.
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Old 12-03-2009, 10:24 AM   #1048
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What he means is that theoretically, if nothing is clipping or distorting, then recording hotter will get you more resolution.

In practice you should rely on your ears AND the peak meter, because things could be clipping at different points in your signal chain before your converters and you'd never know if you weren't paying attention.

Turning down the last gain stage so the converters don't clip isn't going to save you if you're clipping earlier in the signal chain.
Right. Ok, got you. Thanks for removing the fog on this one.
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Old 12-03-2009, 10:50 AM   #1049
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There's also a concept, though, that mixing to 0dBFS is just bad practice -- that, in the analog days, you *never* mixed with your VU meters slammed all the way to the right, even though that was the top of your headroom. The "industry standard" for outboard gear was +4dBu/1.23 volts, which is where they set 0VU. If you do the math in the post you come up with a -20dBFS calibration on the digital scale for for 0VU/+4dBu/1.23V. The argument is that most pro quality plugins use -20dBFS their calibration point, so sending your plugins a 0dBFS signal is like sending your outboard gear a 24dBu signal... In other words, a face-meltingly hot signal. Not a problem for digital plugins, mind you, but for analog modeling this can really change the tone of what you get.
Bang on. But if you ARE recording in 16bit, you'll want to (or, maybe, need to) record hot (but without digital clipping) then turn down the level of the item before any other plugins.
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Old 12-03-2009, 11:14 AM   #1050
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In practice you should rely on your ears AND the peak meter, because things could be clipping at different points in your signal chain before your converters and you'd never know if you weren't paying attention.

Turning down the last gain stage so the converters don't clip isn't going to save you if you're clipping earlier in the signal chain.
This is huge and could be very easily overlooked.
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Old 12-03-2009, 05:38 PM   #1051
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There's also a concept, though, that mixing to 0dBFS is just bad practice -- that, in the analog days, you *never* mixed with your VU meters slammed all the way to the right, even though that was the top of your headroom. The "industry standard" for outboard gear was +4dBu/1.23 volts, which is where they set 0VU. If you do the math in the post you come up with a -20dBFS calibration on the digital scale for for 0VU/+4dBu/1.23V. The argument is that most pro quality plugins use -20dBFS their calibration point, so sending your plugins a 0dBFS signal is like sending your outboard gear a 24dBu signal... In other words, a face-meltingly hot signal. Not a problem for digital plugins, mind you, but for analog modeling this can really change the tone of what you get.

This is confusing because VU is basically an RMS measurement, and for software, dbfs is set to PEAK measurement by default. A plucked acoustic might peak at -3dbfs, but measure -20db RMS. This would match up perfectly with a VU meter in a system that has been calibrated to K-20, (-20dbfs = 0 VU). On the other hand, a super distorted guitar might peak at -3dbfs and measure -7dbfs RMS. This would equal +13db VU on the same system. So, like yep says, there are no formulas, it all depends.

AFAIK, there is no real 'standard' for converting dbfs to Volume Units. Its all up to you and how you decide to align your equipment. I haven't tested my converter's output, but I've always assumed the +4 output it refers to on the back is at 0dbfs (??? I should download the manual and check the specs) ***EDIT: my converter is software selectable for +4dbu output; -15 or -9 dbfs***. As for mixing INB, Reaper is awesome because the master fader's meter shows clearly BOTH peak and RMS. You can solo a recorded track and set the input gain's RMS that way, and see where the peaks are sitting.

The lack of a true standard makes it hell trying to understand what is really going on. All the more important to use one's ears.

Last edited by thinking allowed; 12-10-2009 at 03:29 AM. Reason: clarify
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Old 12-03-2009, 05:39 PM   #1052
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Here's an explanation I read a few years ago online, which I have no way of knowing if its true or not, but I tend to agree with. It goes something like this:

---

Digital full scale at 24 bits = 144db of resolution (theoretically).
In the old *analog* days, the line amps and mic pre's never needed that much resolution. A good preamp might have a -96db noise floor, plus 20db of headroom over unity for peaks (+20db over 0db VU). 96 + 20 = 116. That means 116db of total resolution for this theoretical preamp. Plenty of room for 16 bit (which is 96db of resolution). Way short of 144db for 24bit.

These analog preamp designs were standardized to have their best specs at unity gain (flattest response curve/ lowest distortion figures, best slew rate) and their specs would start to change at levels below, and above, unity gain (0db VU).

Some good designs of the day were turned into little IC chips that could be cheaply made, and easily incorporated into tons of designs. They were very inexpensive, and they sounded great, maybe better with lower noise than a hand built preamp. These designs have worked well and are still made, and still what most of today's audio gear uses.

Using the earlier example of a preamp with 116 db of total resolution:
Lets say I'm building a 24bit converter with IC preamps, and I want ALL the resolution I can get. If I design the analog preamp so that 0dbVU will get converted to digital at -20dbfs, it will allow my full 116db resolution from the IC chip preamp, and put the noise floor at -116dbfs in digital.

But this will mean my best performance from the IC will now occur at -20dbfs RMS. If I record above that, the response changes. This change results in subtle coloration and distortion, the same way an analog preamp does when its being driven too hard. Unfortunately, this isn't as cool as old discrete analog boards being overdriven, the IC's don't act the same.
---

So that's the story, again I don't know if its the truth. We do know that things sound not as right when we record hot. But it would take an electrical engineer and software engineer to figure out if it is our converter's preamp chip, A/D chip, circuit design, or if it is purely software related.

The bottom line: 80db of resolution was more than enough for tape, so yeah, you have nothing to worry about recording lower.

I don't mean to steer off topic, ... I'm eager for more drum info

-----
T.a.

Last edited by thinking allowed; 12-03-2009 at 05:47 PM. Reason: typo
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Old 12-03-2009, 09:48 PM   #1053
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The few recent posts about recording levels, meters and the link given on the gearslutz thread has been extremely illuminating. While I was reading the posts, a light went on, and I opened Reaper to verify. Damn... it was staring me in the face all that time. On the mixer tab, all track faders are at the 0dB unity position. And wouldn't you know it, if you bring down any fader by 0.01dB, it lines up perfectly with the -18dB mark, which happen to be the value given as being pretty much optimal for tracking in this article (-18dBFS = 0dbVU)

http://www.massivemastering.com/blog...ing_Levels.php

I guess the short story is: You don't know at what level to record in Reaper? Then reach for the -18 mark on the track level meter in the mixer tab and you'll be good to go.
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Old 12-04-2009, 06:53 AM   #1054
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Ok so semi-related to the clipping thing, but a bit opposite (and I'm an amature so forgive my ignorance). I'm not clipping my recordings, but rather finding it almost IMPOSSIBLE to clip. In other words, I'm getting nowhere near -12 for a recording of my acoustic without really pushing up the gain and making things sound hissy.

I've been trying to record my acoustic through a Firebox and an AT-2020 (yeah yeah, it's a piece of crap but it's what I got). I don't yet have a preamp other than the ones included in the Firebox so I have to deal with it. If I turn up the gain too much I get a lot of hiss from the mic. I can put a gate on it, but that makes the guitar sound a bit too "harsh" for lack of a better word. So I tried recording a stereo track with the acoustic miked on line 1 and a direct input from my pickup to line 2. I then turned the track into a mono track which actually sounded pretty good with a little EQ.

My problem is that usually...unless I really push up the gain (which increases the noise "floor"...the signal comes in at around -24db which seems very low to me. So I've tried doing things like adding chorus to make it a bit more full and then doubling or even tripling the track, panning to left and right, adding reverb, adding an amp plugin, but with each of these I get farther and farther away from the acoustic sound that I want.

Is -24 through a mic normal for an acoustic? Or should it be louder than that?

I realize that a preamp would allow me to increase gain more, but if I have a good recording that's just quiet what kinds of things can I do to make it a bit louder?
Ok...so after reading the posts after this I realize that -24 is actually pretty close to where I want to be with my acoustic. I think I was misunderstanding the meaning of the term "headroom" (though I don't see why, it's pretty clear).

It's a limitation of my mic and preamps that I can't get the signal hotter (though if -24 is where I'm recording I wouldn't need it much hotter anyhow), so given those limitations, what are some things that I can do to bulk up an acoustic track to make it SOUND louder/fuller?

And a follow on question...what exactly is happening to the audio when I take a track and double it in the sequencer. It gets louder obviously, but is this the same type of amplification that happens when I just adjust the volume of the track?
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Old 12-04-2009, 08:00 AM   #1055
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Yep, as always: my respect and appreciation. I think it's amazing how you can keep this up.

The second reason why I posted is I can't believe anyone reacted on this:

Quote:
Originally Posted by Jonas_Eriksson_Swe View Post
I used to be the bassplayer in a band called Refused which toured extensively across europe and elsewhere. My "trick" for having fresh sounding strings every night was to boil them in water with just a little dish soap in it. This removes some of the fat from the strings (which accounts for part of the "old strings" sound). Although this does *not* make the strings as good as new they regain some of the growl and treble that they used to have when they were new. At least for live use it worked great.

Regards,
- Jonas
Dude, Refused ROCKED. "Shape of Punk to Come" is one of THE defining albums in its genre (and still a blast to listen to), and you guys are owed much respect.

People, if you like intelligent, challenging punk rock and you haven't picked up on these guys: check 'em out.
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Old 12-04-2009, 08:23 AM   #1056
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Ok...so after reading the posts after this I realize that -24 is actually pretty close to where I want to be with my acoustic. I think I was misunderstanding the meaning of the term "headroom" (though I don't see why, it's pretty clear).

And a follow on question...what exactly is happening to the audio when I take a track and double it in the sequencer. It gets louder obviously, but is this the same type of amplification that happens when I just adjust the volume of the track?
maybe i'm wrong but I´m afraid you're confusing ppm (PEAK program meters) and VU / RMS Meters which measure average signal levels. That's why you can adjust how much dB is added to the measured level (don't know the english expression) on VU meters depending on the recorded material. is it peaky but with gaps between the peaks, like drums or superdistorted numötal guitar with NO peaks and NO gaps only constant wuahh etc? a close miced picked acoustic guitar might have higher peaks, ONLY displayed on a ppm, but less average level, diplayed on a VU or RMS meter. a strummed acoustic, not so peaky but more average level.

The term "headroom" always refers to "how much louder until it peaks" -- ppm.
So -24dB on a ppm may be a little low, on a VU or RMS perfectly fine.

when doubling tracks in a sequencer you're basically adding +6dB, in the analog world this equals doubling the voltage. when doing so, you shouldn't use any plugins on the original tracks but route them on an aux or group track and process them there because of phase/latency issues. most sequencers, except for pro tools le (i mean, can they be serious about that?) do have latency compensation but still, be afraid of the phase!!
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Old 12-04-2009, 08:28 AM   #1057
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Yep, as always: my respect and appreciation. I think it's amazing how you can keep this up.

The second reason why I posted is I can't believe anyone reacted on this:



Dude, Refused ROCKED. "Shape of Punk to Come" is one of THE defining albums in its genre (and still a blast to listen to), and you guys are owed much respect.

People, if you like intelligent, challenging punk rock and you haven't picked up on these guys: check 'em out.

true, one of the defining albums of my long gone youth. at that point unbelievable music and still a blast. i don't think there even was a genre for that back then.
i even tried the cook your strings method after i read this. to stings broke when re ah putting them on the bass the two remaining sounded fine.
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Old 12-04-2009, 11:54 AM   #1058
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The bottom line: 80db of resolution was more than enough for tape, so yeah, you have nothing to worry about recording lower.
Ahhh, very enlightening T.a. Thx for that post, didn't realize it was a IC thing. And I like the bottom line, reminded me of the best post from that GS thread. It came from Paul Frindle, one of the original SSL guys and the brains behind Sony Oxford. After going through this ridiculously complicated scenario about inter-sample peaks and behaviors between plugins and how 64bit doesn't guarantee no clipping, he says:

Quote:
I could go on and on but this is getting tedious. Do I think I have covered all major combinations? No not in the least - how can I know all possible combinations of these things - I am only a designer of the stuff which I have designed?

So what about the users - how are they ever to know? How can they ever be other than 'clueless' by definition.

Do you think a block diagram of anything less than a whole book would help you out of this?

So whilst people are struggling away with all this stuff wondering this and fearing that, another engineer has simply given up even trying to understand this shit. He hasn't the time or inclination to figure out what combination of this or that, in this or that order, under these or those conditions, may or may not produce an error he might or might not hear - which his customers may or may not suffer from.

He's put aside the dire fears about the much-dreaded concept of 'resolution' - and simply trims the gain down at the head of every track - to produce something like the headroom analogue system enjoyed (for good reason) - and gets on the with the business of producing art - which is after all, what he's paid to do :-)
Word, word, and word! Enough of this mathy foray into the weeds. Yep, hit us with some drum stuff.
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Old 12-04-2009, 02:39 PM   #1059
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While people are talking about gain staging . . .

I read the epic thread on gearslutz on which the concensus seems to be to record with peaks at 18 - 20dBFS. So I thought I'd try this and recorded a test track of a snare sample, peaking at 18dBFS, then ran it through some different compression plugins. Did the same with a vocal track from my last session, using the item properties to bring it down to the required level (peaks at 18dBFS).

Major Tom, Rocket, Blockfish - no problem but then I tried it through Waves SSL E channel. Cranked up the ratio and started to bring down the threshold only to find that it stops at -20. The only way I was able to get any compression was to use the trim pot to bring the level up to give me something to work with. I was able to use the output fader to bring it back down afterwards. Is this right or am I missing something?
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Old 12-04-2009, 02:57 PM   #1060
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While people are talking about gain staging . . .

I read the epic thread on gearslutz on which the concensus seems to be to record with peaks at 18 - 20dBFS. So I thought I'd try this and recorded a test track of a snare sample, peaking at 18dBFS, then ran it through some different compression plugins. Did the same with a vocal track from my last session, using the item properties to bring it down to the required level (peaks at 18dBFS).

Major Tom, Rocket, Blockfish - no problem but then I tried it through Waves SSL E channel. Cranked up the ratio and started to bring down the threshold only to find that it stops at -20. The only way I was able to get any compression was to use the trim pot to bring the level up to give me something to work with. I was able to use the output fader to bring it back down afterwards. Is this right or am I missing something?
Have you seen the post on the GS thread where a guy sold his Waves plugins? I think it was related to what you're talking about. Ah, here it is:

http://www.gearslutz.com/board/4552991-post131.html
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Old 12-04-2009, 03:38 PM   #1061
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Some recent posters really need to go back and start this thread from the beginning. All this gain-staging, peak vs RMS, and bit-depth stuff was covered at some length earlier. And this thread (sloppy and haphazard as it may be) really does work in a certain order, building from fundamental concepts to more sophisticated ones, that require some more basic understanding.

It will help the flow of the thread considerably if you can quote from earlier posts that you have questions about, rather than re-starting topics that are five pages old. I'd rather not see this turn into a sprawling sub-forum in its own right.
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Old 12-04-2009, 03:41 PM   #1062
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Ok...so after reading the posts after this I realize that -24 is actually pretty close to where I want to be with my acoustic. I think I was misunderstanding the meaning of the term "headroom" (though I don't see why, it's pretty clear).

It's a limitation of my mic and preamps that I can't get the signal hotter (though if -24 is where I'm recording I wouldn't need it much hotter anyhow), so given those limitations, what are some things that I can do to bulk up an acoustic track to make it SOUND louder/fuller?
First off, some of your questions seem to be confusing noise floor with headroom. A good recording needs both-- to push the noise down and preserve headroom. How to achieve this is the entire art and science of audio recording.

If you have not already done so, I'd suggest starting at the very beginning of this thread.

Quote:
And a follow on question...what exactly is happening to the audio when I take a track and double it in the sequencer. It gets louder obviously, but is this the same type of amplification that happens when I just adjust the volume of the track?
Yes, duplicating a track is exactly the same as doubling the gain in mixer (approx a 6dB increase).
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Old 12-04-2009, 04:13 PM   #1063
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Okay, back to drums.

So your job, as the recoding engineer, starts with getting a good kick/snare sound. (When it comes to drums, it's going to become necessary to start with a lot of assumptions, too many to list, because there is simply no way to qualify things to cover every possible scenario, so if anything doesn't apply to you, ignore it, or better yet, post what does apply to you).

I cannot tell you precisely how to get a good kick/snare sound, but they should be developed in tandem-- that is, a kick is only as good as how well it complements the snare, and vice-versa.

Ideally, the drums should have a decay that "dies" well before the next hit happens. This means that tempo and playing style matter a lot. Also, it is important that neither sound overwhelms the other. A huge, booming snare drum might sound awesome, like the fist of God, but if it makes the kick drum sound wimpy by comparison then it's going to cause problems. Similarly, a massive, explosive kick sound that dwarfs the snare accents becomes self-defeating. If all you want is one awesome drum, then skip the kit and just record one drum, seriously.

So the kick and snare should each occupy their own sonic and frequency space (suss this out by ear, please, not spectrum analyzers). Decay time and texture should complement the song and the style of play, and (this is important) should fit into the overall sound and density of the song. Very often, this is somewhat counter-intuitive: a big, dense, roaring arena-rock anthem or slamming club mix often calls for very tight, dry drums, otherwise they may turn into a blurry, indistinct wash that mucks up the whole mix. Similarly, a sparse, spacious ballad or atmospheric piece often calls for big, rolling, textured drum sounds.

Especially if you are limited in terms of gear, room, or instrument selection, drums may require a significant amount of gating, eq, and/or compression just to get the right basic sound. Don't hesitate to do some of this at tracking, to get the drums to sound as close to the way you "wish" they sounded from the get-go. Sometimes processing heavily during tracking will allow you to make better mic placement decisions, especially if the "natural" sound of the drums is not what you need for this track. Feel free to render the tracks with the processing (save a backup project file if you're scared of commitment). It will make mixing a lot easier if you are starting with tracks that are basically "right" to begin with.

Drums and bass should complement each other, sonically. I spent a lot of typing describing how the bass should interact, tempo- and dynamics-wise with the drums. The same is true in reverse. A good recordist will make some decisions about which instruments perform which roles. A pluckier, snappier, and thumpier the bass sound calls for different drum sounds than a rounder, mellower, fatter bass sound, and vice-versa. Which instrument gets priority is a judgement call, but the band's style of play will often make it somewhat obvious. When in doubt, give the drums the "impact" role of thump, snap, and punch, and give the bass the "tone" role of power, fatness, and depth. But let the music lead your decision.

Specifically, know that "thumpy bass" plus "thumpy kick" usually equals LESS overall sense of "thump" and impact, not more. A hundred people stomping their feet has a lot less impact than one foot stomp amplified times 100. "Size" and "punch" tend to be somewhat mutually exclusive. There is nothing wrong with going for a little of both from one instrument, as long as you are aware of and accept the fact that there is kind of a fader that moves between the two: you gain one by rolling back the other.

Hopefully, this sets some wheels turning. More to come.
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Old 12-05-2009, 02:48 AM   #1064
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even snare sounds alone should be developed in tandem, meaning one person listening, the other tilting the snare mic while the drummer is playing. can end up painfull, yes, but is more effective than A/B comparison. it's more fluid ABCDE... comparison.

and one question:
i often read that engineers get the boomyness, the lower mids, of their snare sound from the bottom mic. never managed to do so, always had to eq the top mic. all the bottom mic got was the snare rattle, the snare carpet (?) the thing that snares, you konw what i mean. most of the time i did use a figure 8 mic (ksm44), so less linear frequency response, less bass, but that alone can't be responsable, or does it?
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Old 12-06-2009, 04:51 PM   #1065
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even snare sounds alone should be developed in tandem, meaning one person listening, the other tilting the snare mic while the drummer is playing. ...
Honestly, not just for snare drum, this might be the single best reason to record in a professional environment (or to rely heavily on samples/virtual instruments in a home studio).

The best way to record anything, hands down, is to have a musician playing the instrument, a studio engineer in the control room listening to the sound from the monitors, and an assistant in the live recording space moving the mic around at the instruction of the first engineer, via some kind of talkback system.

Trying to mic a drum kit that you yourself are playing is no easy task, unless you rely on prescribed "recipes", which is potentially problematic for all the reasons talked about earlier (how do you know that such-and-such a recipe is right for your drums, with your tuning, in your room, with your mics? You don't. And it's often not).


Quote:
and one question:
i often read that engineers get the boomyness, the lower mids, of their snare sound from the bottom mic. never managed to do so, always had to eq the top mic. all the bottom mic got was the snare rattle, the snare carpet (?) the thing that snares, you konw what i mean. most of the time i did use a figure 8 mic (ksm44), so less linear frequency response, less bass, but that alone can't be responsable, or does it?
Similar to above, my advice is to forget what you've read about mic placement (or more precisely, remember the underlying reasoning, but forget the specific placements).

If you were recording the same drum kit, in the same room, with the same setup, using the same mics as whatever producer said that, then you'd probably get the same results she gets.

She probably has a collection of four or five snares that she uses regularly, she probably works in a fairly narrow range of genres, she probably hires the same drum tech to do setup for every session, she probably works primarily in one or two familiar studios, and she probably uses the same mic every time, through the same signal chain. Even with all that, when she cites her "favorite technique" or "secret weapon" for recording snare drum, it might be something that she uses 30% of the time, and the rest of the time she does stuff that is completely different.

Moreover, half the places that you read this are probably web forums and magazine articles written by people who are not necessarily recording good records, but just parroting the same "recipe" that they read in this one interview with big-name so-and-so. Do not waste time trying to make recipes "work"-- it either does or it doesn't, and if it doesn't, just try to think about the intent, and find your own mechanics to achieve it. I.e., if the idea is to get one mic to get the "snap" and a second mic to get the "boom", then just find the best place to get the most flattering/useful "boom" rather than trying to force the wrong position to sound a certain way.

Moreover, you have be a bit careful when it comes to advice from recording engineers. They may give a magazine interview where they go to great lengths to explain the million and one options they preserve while recording, and how they like to capture all the textures and sounds, etc (and they may well be very right to do). And then the whole thing might get handed off to a mix engineer who sample-replaces the kick and snare and just uses the overheads and the hi-hat mic, or who only uses half the tracks they've been given to work with. What was captured in the studio does not necessarily make its way to your iPod on that hit record.

Also, if you don't want the rattle of the snare springs, take the snare off the drum. That little lever is there so that drummers can change their sound without changing their drum in a live performance. There is no rule that the springs need to be on the drum if it sounds better without them. Also, be careful of the fact that loose lugs, a bad stand, or other tuning/setup problems can be disguised by the "good" rattle of the snare springs, much as overloaded preamps, pickups, or mics can be disguised by the "good" amp distortion in high-gain guitars.

You might find that you quite like the soft, sizzly "puff" of the snares once the snares and drum are tuned correctly, and once all hardware rattle is gone. Or not. There is a pretty good article on tuning snare drum here:

http://www.musicradar.com/tuition/dr...re-drum-201124

There are a lot more, also. Some others might be more useful.

Drums are a big topic, and can quickly become unmanageable in a context such as this thread, because there are so many variables, from room to mics to inputs to instruments to setup.
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Old 12-09-2009, 05:36 AM   #1066
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Hey yep,

I am guessing that you know of these recordings already, but this site http://www.ricksuchow.com has some great isolated bass tracks from Jamerson's recordings. It's an awesome listen.

Cheers and thanks for all your efforts.

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Old 12-09-2009, 07:09 PM   #1067
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Hey yep,

I am guessing that you know of these recordings already, but this site http://www.ricksuchow.com has some great isolated bass tracks from Jamerson's recordings. It's an awesome listen.

Cheers and thanks for all your efforts.

kindafishy
Cool, I did not know that!
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Old 12-11-2009, 03:39 PM   #1068
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Yep, For three days I've spent every spare minute reading this thread from the PDF file (300 freakin' pages). I've got 10 pages of notes and a new enthusiasm for continuing my recordings. Thanks.

I record our all-acoustic blugrass-ish, band which consists of upright bass, guitar, dobro, mandolin, fiddle and female vocalist w/harmonies. I couldn't agree more that the quality of the "final" mixes I've done is exponentially improved by well-adjusted, well-tuned instruments, attention to mic placement, and rehersal to the "stage 3" level. And, to me, nothing is more important than having musicians with good timing. I think the next thousand dollars most folks need to spend is on lessons and a metronome - they can find the time for this by cutting down on web-surfing for the world's best equipment.

Anyway, our music, it seems to me, calls for minimal "effects", and more focus on "clean space" for each instrument. I've already screwed around with EQ filters based on your advice, and you've got me smiling (because it sounds better and I know why). I'd like to hear your thoughts on placing instruments in "space" - front to back and right and left (I don't think I missed that earlier, but maybe so).

Also, I've had the experience of putting together a mix that sounds good on the monitors and "like ass" in the car - muddy, with, I suspect, the acoustic guitar and bass stepping all over each other in the lower frequency ranges (maybe dobro as well) - also no-"thump" in the bass. Part of this might be what I'm hearing in the mixing room (I'm ordering bass traps next). The EQ tips should help here, but this brings me to the whole acoustic bass/rhythm issue -

Our music has no drum (sorry to sidetrack the recent drum posts). Our rhythm section consists of an upright bass note on the downbeat and a mandolin "chop" on the off-beat. Being "Reaper specific", have you got some ideas how I can draw out the thump on the bass with a tight percussive "chop" from the Mandolin and not have it all become muddy when I stick an acoustic guitar in there? I'm going to play with the compression now that I understand it better, and as noted, the EQ (specifically high-pass and mud-zone) tips have already helped. I know, the first rule is to use my ears - and I'm trying.

Thanks.







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Old 12-11-2009, 06:51 PM   #1069
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...I'd like to hear your thoughts on placing instruments in "space" - front to back and right and left (I don't think I missed that earlier, but maybe so)...
No, I haven't talked about this too much (thanks for the kind words, BTW).

Before I get started on that, an aside on building a sense of rhythm, punch and thump from non-percussion instruments: First, foot-tapping and hand-clapping can be just as good as a drum kit. Second, if a track needs drums then it needs drums. I can no more tell you how to get a string bass and a mandolin to sounds like a kick and snare than I could tell a drummer how to get a kick and snare to sound like a string bass and mandolin. Third, if the music doesn't call for drums, but could still use a little more "punch" or "oomph", look to arrangement and performance first. An ounce of a band that sounds right in the room is worth a metric ton of recording and mixing technique. That said, all the stuff said about guitar and electric bass and everything else is relatively applicable to mandolin and string bass, vis-a-vis compression, frequency, mic placement, and all the rest of it.
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Old 12-11-2009, 07:36 PM   #1070
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In terms of front-to-back space, this is actually a topic that I am a little skeptical of (or maybe just incompetent about). I usually tend to think that good "front-to-back" placement is really no different than just plain old good sound. I.e., an instrument that sounds "too forward" or inappropriately "pushed back" is not much different than an instrument that has been played, recorded, or processed inappropriately.

In a general sense, resonant low mids sound "further back" and dry or gated upper mids sound "up front". And it's often that simple. There is a strong instrument-specific component to this principle. A string bass with a little bit of gating or close-miking, and a smidgen of EQ to bring up the string/finger sound will seem to be pushed a lot closer to the listener, whereas a mandolin with a little reverb and a slight upper-mid rolloff will sound pushed to the other side of the room.

I.e., it's not about how the frequency spectrum looks in an analyzer, it's about the psycho-acoustic cues that tell us the difference between, say, snapping fingers right next to your ears versus snapping your fingers with your arm stretched far away (try it-- it's obvious). One "pops" and even leaves a bit of sharp, residual pain, and we can hear detail and texture in both the high and low frequencies. The other sounds cluckier and thuddier.

Close to the ear, the highs reveal every rasp of the fingerprints in an articulate and detailed way. Further away, they become a ghostly whisp of air. Close in, the lows are a quick, thunky "punch" of pressure on the eardrum. Further away, the lows are are felty, thuddy "puh". Close in, the mids sound like a sharp, slightly painful "click", like a wood block hit sharply by a drumstick. Further away, the mids are a meatier "snap", like the sound of slapping a steak on the counter, if that makes any sense.

In a way, you can learn all you need to know by snapping your fingers or clapping your hands in different ways in different places.

"close" (and "small") sounds like: articulated, textured highs; punchy, thunky lows; and sharp, pointed mids. "far" (and "big") sounds like airy, ghostly, harmonic highs, poofy or wahwy lows, and "bamphy", explosive mids.

Note that I linked close and "small", along with far and "big" sounds. Rolling thunder does not sound "in your face". A lover sighing in your ear or kissing your cheek is about as "close" and "in your face" as sound gets, but it does not sound "big". For men who shave, or for women who apply makeup, that distinctive, articulated rasp and textured brushy sound are the best comparison for a literally "in your face" sound that you can get.

"close" sounds have a dry, textured, sandy, maybe grainy or powdery quality. "far" sounds have a boomy, rolling, "full" and "washed out" quality.

A lot of modern production aims for both "big" and "in your face" from the same sounds. This is a very dramatic, very artificial kind of soundscape that can become rapidly fatiguing to listen to. But good or ill, it's worth talking about.

Maybe the most common example of "big" and "in your face" sound is the stereotypical movie sound effect of a puff on a cigarette. I think everyone is familiar with that throaty, crackly, satisfying sound of burning tobacco in a movie soundtrack. As a former longtime smoker, I can tell you that in reality, a cigarette makes practically zero sound. But the movie sound does a brilliant job of approximating what it FEELS like to take a drag off a cigarette. If all you knew of smoking came from the movies, you'd think that every puff of a cigarette filled the room with a rumbling crunch of crackling tobacco. Another similar effect from the movies is the big, satisfying crunch of boots walking in snow or gravel. Nevermind that it doesn't actually sound like that in real life, it FEELS that way--the skill of the foley artist lies in creating a soundscape that sounds realer than real, larger than life, and truly cinematic.

You can apply similar techniques in music recording, although much like in foley recording, intensity needs to be used with discretion. Making everything sound huge and in-your-face creates a painful, deafening, "TV commercial" sound at any volume.

Audio engineers and audiophiles often make a lot of fuss over "smooth" and "vintage" sounds, and they are frankly right, most of the time. There are not very many popular recordings made in the past 10 years that I expect people will still be listening to in 40 years. Music mixed to sound like a television commercial does not age well.

That said, it would be disingenuous to pretend that such soundscapes and techniques are irrelevant or illegitimate-- they are the dominant sounds on commercial radio today, for good or for ill.

What defines this modern sound of having both size and immediacy, both depth and in-your-face-ness is the combination of textured, detailed mids and highs with depressed, clean lower mids, and deep, booming, exaggerated lows. This is typically achieved by either multitracking or parallel processing (ie, two or more tracks of the same instrument, each track having a different sonic role).

If you have followed this thread all the way through, you should have a pretty good idea of how to achieve these different effects. It bears repeating that recording a single source with multiple mics, or processing it differently across multiple stems creates an unnatural sound, and that unnatural sounds can become tiring, gimmicky, and fatiguing to listen to.
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Old 12-12-2009, 03:11 AM   #1071
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Hey!

I wrote this reply on another thread few days ago or so, but when reading the latest posts here and especially Yep's latest post, I thought this might be a good addition to this thread/conversation too. Oh, and excuse my english, I'm not a native.

So here we go:


Most of it is psychoacoustics. You're trying to fool your ears and more importantly your brain to perceive the summed stereosignal as a real world sonic image - at least to some extent. You can do this by using the information all around you - the nature and its laws of physics. Those are the things your brain can, and will, interpret correctly and reliably. However, this said, most of the times in contemporary music and audio production you're actually required to bend the laws of physics; make things sound larger than life or otherwise manipulate physics of audio in all kinds of not so realistic ways (e.g. in a movie make the dialogue and a gunshot approximately as loud but still retain the impression that the gunshot would have been way way louder than the dialogue was).

First of all, start to listen to all things around you differently and from another perspective - teach your ears to perceive the aural information in a more analytical way. Pay attention to your surroundings and how they sound like. How does the frequency balance of sound behave in correlation with distance? How the dynamics behave? Air compresses signals, but how much? How do I recreate this with compressor - can I? How about reverbs? Whats the purpose of the pre-delay and how it relates to real world around us? How does the reverb work in a horizontal axis? What makes the reverb work in nature? How much reverb do you really need in order to hear it as a spatial information? Is the whole frequency spectrum of the reverb needed for it to work? Are plain delays enough? And so on and so on..

The overall perceived depth in a mix derives itself from many small things: Volume, EQ, Dynamics (compression), reverb(s) and so on. It's a very very very delicate balance between all of these. Also, don't forget one of the basic neccesities: the contrast. If you want to make something sound distant it helps immensely if you have something to hear it against to - more spesific, something wich sounds like it would be located at very close to the listener.

As for some practial examples lets imagine a person talking in a hall at few different locations. First he is talking right into your ear; how does it sound like? If you think about it, you'll notice that there are next to none air in between your ear and the source of the voice. This (amongst other things) causes you to hear the spoken words very clearly, sharp and crispy. Air does't have time to compress the sound nor can it disturb the frequency spectrum. However, even when he is talking directly into your ear you're still able to hear the hall too. The reverb is quite low in volume compared to the voice speaking into your ear, but nevertheless it's there. Also notice the frequncy balance of the reverberating voice compared against the close voice - they're drastically different (super sharp and crispy against muffled midrangey reverb). Last, but definitely not least, one of the most important factor: the reverb of the hall travels into your ears waaaay after the dry signal (pre-delay). Naturally, we can also hear other reflections coming from your cheeck and shoulders and his face and so on, wich will cause your brain to be absolutely sure of the fact that someone is indeed talking right into your ear.

Ok lets move the talking dude further. Now he is talking 20 meters away from you and at the center of the hall or so. Now you hear the effects of air in between you. The highest highs and lowest lows aren't really audible anymore. The voice is also a tad more compressed. The time difference between the dry and reverberated signal is shorter too or reach your ears pretty much at the same time.

Now lets put him beside a wall, quite close in front of you. He talks in front you you but the reverb isn't anymore symmetrically around him or "behind" him. Instead the reverb reverberates from "side" - yet you have no uncertainty of the location of the talker. He is still definitely in A) a hall B) quite close in front of you. What do we learn from this? It's completely OK to e.g. pan the dry signal left and reverberated signal right - brains can interpret nevertheless.

Those were some basic examples of the acoustics around us. It may sound like a cake, but in practise it's a bit more harder to pull off. But as you can, hopefully, see it's a complex balance of EQ, reverb(s)/delay(s) and compression/volume. Also if you understand this, you will see why there can't be any specific presets for this kind of manipulation. One thing might work on a close miced electric guitar but it would be total disaster on a source wich has a more natural ambience in it.

Perceived 3D depth is one of the most hardest things to incorporate into your mixes (imho) and it will take the time and practise to do succesfully and intentionally. It requires the understanding of frequency spectrum (EQ), dynamics (compressor/volume relations) and time based effects (reverb/delay) and their relation to psychoacoustics. And it gets incresily harder when you have to work with different music genres, lots of tracks, artistic strives and so on. Just imagine the situation when you have 40-50 tracks in all varieties and you have to mix them together. Dynamics, "traditional" frequency balance, artistic side of it, effects and so on - and then you have to try to incorporate the spatial imaging into there too? You run into all kinds of problems, especially with reverbs. Like e.g. you want to something to be really lush and reverberated and far away. Ok so you crank up the reverb and sure your track did go far and distant. But the side effect of it is that now the mix is all muffled because you brought a gigantic hall into the mix wich masks other instruments and eats away the clarity. How about the topic of bending physics then? You want something to sound super close and clear and at the same time like it would be in a some sort of superhall? In nature these kinds of phenomena don't really exist. One example of this kind of problem might be a snare drum wich has to stay loud, clear and punchy but at the same time large and massive and reverberated - in other words "larger than a life".

Yours,
Kainz
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Old 12-13-2009, 08:01 AM   #1072
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Well Kainz,
Nicely put set of observations.

It really lead me to one question...

"So then what is/are the list of tricks that pro mixers use to pull of these illusions?"

I think that is what most folks are actually looking for when they see a thread titled like this one.

E.g. "How do I make my tracks sound as big as [band x's] song [fill in song name].

One example I like to use is Blink 182's "All the Small Things". On the surface just seems like it should be a few guitar tracks... but even the pick scrapes sound huge.

Another example, Anything on the first two Van Halen albums. These seem to break the mold that "to sound huge you need to 2, 4 or 8 track the rhythm guitars"... It's the usual example when I pull out when someone says, "oh to get that wall of guitar sounds, double or quadruple track". Well... oh yeah? Then what was EVH's trick???

Yet another example. (Of course there a bazillions) The Metallica Seattle '89 live tracks. The mixing and guitar tones on that DVD actually sound better than the studio albums. (As do AC/DC Live, Live at Leeds, and some tracks on All the World's a Stage.)

Anyway, on one "How to Mix" DVD I saw a part of once, there was literally one track for each drum, plus overheads, 8 guitar tracks, 2 or 3 "lead vocal" tracks and I guess there were 40 plugins being used, best estimate. I heard the guy say 3 reverbs and 6 delays were part of his basic template. In the end, every single thing you would have thought of as a discrete "sound" or "track" was really a composition of layers, and layers and layers... There was literally *nothing* that wasn't f-ed with in some way.

Geezus, no wonder when I was in my 20's I couldn't "Find the Sound" with my guitar and amp. Could never "sound like the record" Because what was on the record, never existing on the front side of the mic. (Heck, I ended up running 2 or 3 different amps in parallel, just to get close. And still to this day, as a musician, feel completely lied to and fooled by the musical equipment industry, or thier ad writers anyway. All the BS about "this stomp or that pickup is the secret to sound x... ya, right, and a cabinet full of mics, and a studio full of tools.)

So moving on, besides gain staging there's "huge sounding in genre x" to deal with.

With, in the world of pop and classic rock that I tend to follow, there seems to be two styles:

1) Bazillions of Tracks, highly engineered with layering all over the place for every sound.

2) What I'll call "Power Trio et. al."... in short the exceptions like EVH and perhaps ZZ Top.

As for me I continue to look for the trick to #2. Having started as a guitar player I want that apparent ease that EVH and Billy Gibbons seem to have to just sit down, blast out the guitar track and actually be playing and hearing the sound that will end up on the recording.

Unfortunately that seems to be rare. Most chances I've had to examine what a pro mixer does, it's #1. The number of tracks, amount of tricks, etc. pretty much says that a modern recording is more like using recording tools to edit recordings the same way a word processor is used to edit a book.

100% thought out, reworked, re-edited, engineered, and so on. In short, you'll never get to play the sound on the recording, because it was never a discrete event like in the earlier days of recording, where performing artists came in, were expected to play the songs straight through and the idea as to capture a great performance.

The deal since multitracking has occurred is the "construction" of recordings. Heck you could easily construct a recording and have never played the song "all the way through". (Good thing too, for me, I'll never have time in this life to be the old style performer/player.)

So anyway, the bottom line point is this. What we're really all looking for, in my estimation is:

"The Bible of Studio Tricks: How the engineers made average sounds, sound huge"...

As an example, go over to Youtube and find the "The Story of Bohemian Rhapsody" series of videos. It's about a 6 video set.

In sections 3 and 4 you get to hear Brian May blasting in the studio...

Well, guess what... the sound the camera mic picked up sounded like any other garage band sounding guitar and amp.

So therein lies the rub... How *did* the engineers (and composers) get it to all sound huge, given that each individual track is almost as awful as what I hear at Guitar Center on any given day.

And *that* I believe covers a lot more ground than the effects of gain staging...
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Old 12-13-2009, 08:59 AM   #1073
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While it's clear that some mixes are done that way, I think a lot of great ones can fall under your #2 category. Using the tools judiciously, you can create that gunshot sound without octo-layering tracks.
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Old 12-13-2009, 09:23 AM   #1074
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"So then what is/are the list of tricks that pro mixers use to pull of these illusions?"

...
There really aren't any specific "tricks" or magic to turn a mix into a "pro sounding". Honestly, there isn't. Things like that just don't exist. Of course there are some mixing techniques wich could be perhaps labeled as "tricks", but I don't really see it that way - as of them being "tricks" I mean. These techniques include commonly used things like paraller compression, volume automation, clever delay usage (e.g. producing Haas effect) etc. But none of those "tricks" will turn a bad/mediocre mix suddenly into a "pro sounding" mix. They're just tools and tools don't do anything by itself. You need to use them properly and well in order to create something sensible with them.

All answers for your questions are really already existing in this tread: Make sure your tracks sound as a record to begin with; good rooms/spaces, mics and a lot of critical listening are worth more than any "mixing tricks". Never "fix in a mix". Make sure you take care of editing properly and clean up all unneccesary stuff (this doesn't mean autotune and/or beat detective). After you got these right, mix the song as you see fit. And remember that every mix is different, totally different. Why? Because every song is different, tracks are different, everything is different. There are no "presets".

The skill of mixing is a labour wich requires a lot of practise to master. It just won't work so that you read few books/watch some "tutorial videos" and suddenly be a superb audioengineer. Nope, it takes usually years to be able to mix well. The paradoxical part of this is that even now when you listen to your own tracks/mixes you hear exactly the same things than what a "pro engineer" would hear. The difference is that you don't realize you hear various problems in it but the pro guy does, thus enabling him to fix and/or change things. The art of mixing is really an art of listening; learning to listen in various ways and various things. Like Yep but it in his post: "It's like that thing in the Bible: they listen, but they do not hear."

I mean the pro guy has exactly same stuff as you have: EQs, compressors, reverbs, delays and bunch of other effects. Nothing mystical in it. Of course he might have some quality hardware wich home recordist might have not, but that isn't really an valid argument nowadays. The difference in gear (physical or virtual) isn't by any means large enough to explain the difference between a "pro sound" and a "home sound". One thing wich comes to my mind, however, is the difference in automation usage between a pro and a hobbyist. In a "pro mix" there are tons of automation going on, almost every track is somehow automated. Why? To keep things moving and interesting for the listener! A mix can't just "be" it needs to move and be exciting and support the song in a best possible way.

Quote:
I think that is what most folks are actually looking for when they see a thread titled like this one.

E.g. "How do I make my tracks sound as big as [band x's] song [fill in song name].
First of all this whole thread is a comprehensive and elaborate answer into this specific question. The thing and the truth is that there isn't one answer to rule them all. It's a sum of many many many many many small things wich as a sum make the more or less drastic difference between a "pro sound" and a "home sound".

Another thing is that no one can tell you how to listen. It just doesn't work like that. Of course someone can help you with it by pointing out various things in a mix, but eventually you just have to teach your ears by yourself. It all culminates into this; mixing, recording and all of it - into hearing and listening, I mean. Just think about it for a second: If you knew what makes your mixes suck wouldn't you just fix it? So what prevents you from doing this? The one simple fact that you can't hear it - you listen to it but you don't realize it. It really is as simple as that. As I wrote earlier the difference between a "pro" and a "hobbyist" is mainly and most importantly the fact that the pro can hear it whereas the hobbyist cannot.

Thats the whole core of mixing: To hear the problems and solve them. In the beginning your problems are relatively large and obvious (wich most beginners can spot without a problem also), like e.g. the guitar is 10 times louder than the vocal track. When your mix progresses you move towards smaller and smaller problems. Somewhere in this process the beginner can't hear the problems anymore whereas the pro can continue forward. And eventually the problems are so minuscule that it isn't meaningful to fix them anymore and at that point the mix is starting to be ready.

Last edited by Kainz; 12-14-2009 at 07:51 AM.
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Old 12-13-2009, 10:07 AM   #1075
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Thanks Yep and Kainz, for taking the time to help me understand this.

It makes perfect sense that the sound of a track or instrument will sound "far" or "near", "big" or "small", based the multiple interdependent variables you described. So, I have a good foundation for some trial and error work.

I'm sure flmason is correct, that some want to hear the top 10 tricks for big sound, but, I'm not ready for that. I'd want to know why these tricks work, because they're unlikely to apply to my <10 tracks- no drums, no-synths, acoustic mixes. Hell, I'm still working to understand "clean and clear" and hoping to get "fuller" without screwing that up.

You didn't address panning, perhaps because it's only a matter of taste. In my mixes, I pick an instrument and stick it a little right, then pick one (usually one that operates near the same frequency range, like dobro and guitar) and pan it an equal amount left. I center the lead vocal and offsett R/L the harmonies (typically 2 other vocals). I stick the bass right in the middle. I have no rationale for this... only that it seems to sound alright to me, and much better than all-center. Should I be listening for anything specific here?

To now, my only "front to back" has been volume control, and reverb/delay differences by instrument. But I've got new ideas from thinking about snapping fingers and a dude in a hallway.

PS: Yep, I know you're good, but even I didn't expect you to be able to tell me how to make a bass and mandolin sound like a drum in the mix!
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Old 12-13-2009, 12:42 PM   #1076
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I heard the guy say 3 reverbs and 6 delays were part of his basic template.
I think you may be interpreting this - and statements like this - in the wrong (for lack of a better term) way. You're looking for 3 reverbs and 6 delays to be a formula for a good mix. More likely this was a professional mixing engineer who after years of trial and error and access to thousands of plugins and millions of dollars worth of hardware, has found that he can do most of what he wants with just 3 reverbs and 6 delays.

I think you're hearing this and thinking "3 verbs, 6 delays! So many things going on! How am I supposed to make sense of all that?" But this engineer didn't just wake up one morning and think "From now on I'm going to use 3 verbs and 6 delays on everything!" What more likely happened is that he found that mix after mix he wound up using 3 verbs and 6 delays to get the sound he wants. So to save time he loads them up as a basic template

And if you break it down, six delays really isn't very many. Especially if any or all of them are being used for stereo widening, then they're being used in pairs anyway; e.g., source in center, delay on the left, slightly different delay on the right. Three reverbs is not an excessive number at all: 1 for vocals, 1 for the snare and one to create the "room."

Experience will teach you. Don't be afraid to play. Take a song you think you mixed pretty well and wipe out all the plugins and start from the beginning. Too dry sounding with no plugins? Try adding a room reverb.....
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Old 12-14-2009, 01:02 AM   #1077
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I think you may be interpreting this - and statements like this - in the wrong (for lack of a better term) way. You're looking for 3 reverbs and 6 delays to be a formula for a good mix. More likely this was a professional mixing engineer who after years of trial and error and access to thousands of plugins and millions of dollars worth of hardware, has found that he can do most of what he wants with just 3 reverbs and 6 delays.

I think you're hearing this and thinking "3 verbs, 6 delays! So many things going on! How am I supposed to make sense of all that?" But this engineer didn't just wake up one morning and think "From now on I'm going to use 3 verbs and 6 delays on everything!" What more likely happened is that he found that mix after mix he wound up using 3 verbs and 6 delays to get the sound he wants. So to save time he loads them up as a basic template

And if you break it down, six delays really isn't very many. Especially if any or all of them are being used for stereo widening, then they're being used in pairs anyway; e.g., source in center, delay on the left, slightly different delay on the right. Three reverbs is not an excessive number at all: 1 for vocals, 1 for the snare and one to create the "room."

Experience will teach you. Don't be afraid to play. Take a song you think you mixed pretty well and wipe out all the plugins and start from the beginning. Too dry sounding with no plugins? Try adding a room reverb.....
No I'm not misinterpreting at all. I concur that said engineer did a lot of study and experiment to get to his template.

My point is, back 20-30 years ago, as a budding guitar player... what I thought was the sound of "pro electic guitar" as heard on the classic records of the day (which obviously began in the pre-plugin days) was *never* what you heard in the room with a guitar and amp. And often the amps of the pros were modified, and they had access to things you didn't. And of course the studio and engineering.

Point being, can't say how much money I blew on Marshalls, Fenders, Ampegs, stomps, etc. looking for "the sound". When in fact "the sound" of the recording may have never actually been played by the musicians.

I found that realization to be a let down, for sure.

It also meant that the cost of making a pro sounding recording, or a recording-like sounding live show just went up, LOL!

I stepped out of it for about 10-15 years. Saw Cubase and AmpFarm in "Home Recording" back around 2000-ish and decided to investigate, on the theory that it could all be done in the box, even without amps. I read that Amy Mann or some such did here entire album at the time ITB.

That seemed like a good thing since my day career had repeatedly thown my life into turmoil. No way to keep moving all the amps around I had back in the analog days.

So... tried out POD 2.3 and the Brit High Gain patch + delay, reverb and chorus put a big grin on my face...though oddly, no Line 6 sim since has. And they don't make a POD 2.3 plugin.

Anyway, what I *still* found out is, even with magical digital stuff like the Pros who use AmpFarm and ProTools you don't magically end up with a pro, or even a lifelike sounding result.

And to add to the aggravation, every amp sim maker, but *especially* Line 6 wrote all their manuals and prose like, "Hey just turn on the Super-Pro sim and you'll be Rockin' in the Holy House of tone"... (e.g. Led Zep tone)... "instantly".

Yeah, what a crock that is.

So the point of the above posting pretty much remains.

The reason many of our mixes sound like ass is because we don't know all the accumulated studio tricks of the last 40, 50, 60 years of rock and pop recording. All the illusions and how they are done.

And *that's* probably where this thread needs to go. To a compendium of what are considered to be standard solutions to typical problems in both mixing *and* composition.

Composition? Yes! I have auditioned many midi tracks on the web through soundfonts... if the midi is accurate to the song, you can get a near pro sound out of just soundfonts! I.e. the composition is where it all starts. The orchestration.

But... those soundfonts I've used are idealized... each note perfectly compressed to be equal volume with the others... etc.

So that leads to tracking...

and then to mixing...

At tracking and later at mixing and effecting, you have to know what sound you are shooting for, for the track and how to get there. And as many have pointed out, it's not always what sounds good in solo. Nor is it always just *one* track...

Seems literally each instrument, and each genre has a history of studio tricks that lead to sounds we now call "Classic" and *thats* the tricks most are *really* looking for. *That's* the "Pro Sound" everyone is talking about, and cares about. Despite all the back and forth about "bit depths" etc. That's all just more "Tone Mojo BS" that goes back the days before you had something like ReaFir that could surgically modify something.

When all you had was a Neve 1081 channel... well yeah, maybe a really flat mic like a U87 *was* the trick, because it played nice with the 1081. So a tone mojo tradition got started. But in the end, that low cut on the U87... is a big part of getting some vocals to stand out in a dense rock mix...

Or another example, perhaps the use of Celestion T-75 speakers to "cut through" a dense mix. Well... if you know how a T-75 speaker's response curve is different from an Oxford 12L6... perhaps that can tell you what you can do with EQ to make *any* speaker you're stuck with, cut through the mix.

So if you know the *real* story and the *real* effect you are looking for... you can probably approach it with significantly flexible tools.

And... if you tools are that flexible, come up with something new. Wherein the "classic" sounds are just one possible set of settings. (But if you want the "classic" sound, you still have to know what the various curves look like, etc.)

So to sum up. What most of us dummies are probably really looking for are descriptions and examples of the tricks and techniques that *work*. The ones that when you flip them on, you go, "YEAH THAT'S IT".

And perhaps for those of us on a budget, *why* those things work, so we can approach on them with any tools we may have on hand.

In short, the next step is making a science out of it, rather than a bunch of Tone Mojo Mythology.

I believe that starts with a thorough cataloging of "standard" or "well known in the industry" methods to be used as case studies and instruction.
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Old 12-14-2009, 07:11 AM   #1078
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Originally Posted by flmason View Post
My point is, back 20-30 years ago, as a budding guitar player... what I thought was the sound of "pro electic guitar" as heard on the classic records of the day (which obviously began in the pre-plugin days) was *never* what you heard in the room with a guitar and amp.

Point being, can't say how much money I blew on Marshalls, Fenders, Ampegs, stomps, etc. looking for "the sound". When in fact "the sound" of the recording may have never actually been played by the musicians.
I'm not completely following this logic here. I mean, how did they developed their sounds in the first place? They most likely listened to some of the guys/idols of the time (e.g. Hendrix) and took some influences and so on, but eventually they ended up with their own sound. For example Yngwie definitely has his own sound as does mr. Eddie. And, you know, they did figure out their sounds before going to the studio. The studio didn't invent their sounds, the players did.

I'm saying that most of the time the recording engineer is asked to reproduce (record) the sound the artist is hearing from their amp instead of trying to "reinvent" it. This is still one of the basic qustions you have to ask from a guitarist when you're recording him: "Is this your sound coming out from the speakers?" If the answer is no, there is no point recording that and you're back at the "drawing board"; move mics around, test different mics and/or micing techniques etc. And belive me, guitarists are generally very strict in this matter. They want to hear their sound, nothing more or nothing less.

But FYI most of the "rock/heavy" guitar sounds back in the days were recorded with 1-3 mics (and still are). The most used mics were the SM57 and the MD421. There are various mic placement techniques they used, but in general the cabinet was always close miced. Meaning the mic was basically touching the grill of the cabinet pointing straight into the element from various angles. There might have been some EQ applied after the pre-amp before going to tape (mainly to cut some lows/clean up the low end so that the tape wouldn't distort unneccesarely) and most likely some sort of protective compressor/limiter to prevent, again, overdriving the tape too hard. In the mixing stage some delays and/or reverb(s) might have been added with some basic EQ to fit the tracks together as a whole. Thats about it, nothing too magial in it, right?

Quote:
The reason many of our mixes sound like ass is because we don't know all the accumulated studio tricks of the last 40, 50, 60 years of rock and pop recording. All the illusions and how they are done.

And *that's* probably where this thread needs to go. To a compendium of what are considered to be standard solutions to typical problems in both mixing *and* composition.
Oh man, did you read my earlier post at all where I tried to explain this to you? Try to understand this: Most of the times recording engineer's job is nothing but A) To record a specific instrument/sound as we're hearing it as well as possible. B) To fix various problems in mixing/recording stage wich are not beneficial on the end result. C) To make sure everything is heard in a mix and the mix supports the song as a whole as well as possible (the creative side of it).

One doesn't think about "tricks" or something like that when you record and mix. You just listen for problems and then figure out various things you could use to solve these problems. For example, "The guitar has too much low end in it" -> Fix: "I'll use EQ to cut it out". Or something like, "This background vocal is too dynamic and I can't find a good static fader position for it due to that" -> Fix: "I'll use either volume automation or compressor to control the dynamics in it". Thats the core of it, listen and fix - all over again.

Some of the ways developed to solve all these problems might have been "elevated" into a status of being some sort of "tricks" or maby tied into a sound of a specific musical genre. Like for example doubled or quadrupled heavy rythm guitars in a modern heavy music. That could be labeled as a "trick" and it's quite hard to get that modern heavy sound without using this kind of recording method when creating the soundscape. But still, there is nothing "magical" in it really. We just needed a heavier sound thus we duplicated things (a problem and a fix). It's also a good idea to remember that this "trick" won't turn sucky guitar sound into a "pro sound" no matter how many tracks you lay down or (ab)use this "trick". All these thins are just tools, and you have to know how and why to use them. It's like e.g. a language; you could memorize all the words from a foreign language but yet you wouldn't be able to speak it. It's no use to try tell the answers if one isn't able to ask (hear) the questions. I'm really out of ideas how to elaborate this more. There are no "standard solutions to typical problems". If it would be that easy, don't you think everyone would be a world class mixing engineer? Just buy the allmighty Compendium of Problems and Solutions 4th edition and be done with it. Nope, not really.

First of all, there are nothing but a handful of "typical problems" wich we could carry over from a mix to another one. You do realize that in each song and in each mix all tracks are different? You can't apply any sort of "preset" into them and suddenly they would sound like the best thing you've ever heard. Currently you're trying to explaing things in a way wich can't exist. You're kind of trying to find a some sort of englightening shortcut to all of it, but frankly one doesn't exist. It's all about practise practise and more practise.

Quote:
When all you had was a Neve 1081 channel... well yeah, maybe a really flat mic like a U87 *was* the trick, because it played nice with the 1081. So a tone mojo tradition got started. But in the end, that low cut on the U87... is a big part of getting some vocals to stand out in a dense rock mix...

Or another example, perhaps the use of Celestion T-75 speakers to "cut through" a dense mix. Well... if you know how a T-75 speaker's response curve is different from an Oxford 12L6... perhaps that can tell you what you can do with EQ to make *any* speaker you're stuck with, cut through the mix.
Ok, let me put it this way (bit "romanticized" but nevertheless). Wich one do you think was first, the problem or the solution? Did the engineer record with U87 and then few years later decide that "damn this low cut does the trick". Nope, it was other way around. There was a problem "We need this vocal to cut throught" and they ended up using the U87 because it sounded the most what they were looking for. He LISTENED the vocal, spot a problem with it and found a solution for it. He didn't think: "Hmh, what did the guys use at the 70's to get this vocal sound? Oh man I can't make this track "cut" unless I know it!"

You might be facing a similar problem with your mixes, who knows, and your job is to solve it. Does it require the U87? - definitely not! What does it require then? It requires the ability to hear the problem and use the appropriate tools to solve it. The point is that you have to listen first; always listen and identify before touching anything. That way you will also understand the aspect of WHY in it. Currently it's like you would be trying to fix a car in a pitch-black room and you're moaning that you can't fix it because you don't have all these superb tools and the know-how of a pro mechanic. I can assure you that the pro mechanic wouldn't be able to do it in that pitch-black room either; even with all his tools and know-hows with him! Turn on the lights - listen to it! - and look around, see (isten) where the problem is and think about wich tool to use and why. Use a screwdriver to tighten a loose screw - or use a compressor to tighten up a loose track. The thing is, you have to be able to spot the problem first, only then you can apply ANY tools or "tricks" into it.

And what it comes to your speaker example. There is an EQ invented to boost/cut things. You use it by listening and then turning the knob(s), simple as that. Your analogue of various speaker "response curves" sounds a bit alien to me. I'd be mad if I'd be thinking things like that when I mix. I just listen and turn the knob(s) until I hear what I want (and most of the times I know what I want to hear before I turn anything). Besides, lets say we have a cabined loaded with 12L6. Then we record if with SM57 and then with MD421 and then with U87. Are you saying that all I need to know is how the T-75 "response curve is" and I can apply a some sort of "preset" into all of these 3 tracks to get them to "cut in the mix"?. That just would not work at all. Way way way easier solution would be to just to listen the mix as a whole and boost few things in the guitar track(s) and be done with it. Who cares what frequences those are as long as they do the trick? This whole "cut throught the mix" concept is also totally dependent from all the rest tracks in any particular mix. Even if I boosted the exact same area(s) it wouldn't work in all mixes - it could "cut" in one but wouldn't do it in another.

Continues...

Last edited by Kainz; 12-14-2009 at 08:16 AM.
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Old 12-14-2009, 07:12 AM   #1079
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... continues.

Quote:
So to sum up. What most of us dummies are probably really looking for are descriptions and examples of the tricks and techniques that *work*. The ones that when you flip them on, you go, "YEAH THAT'S IT".
...
I believe that starts with a thorough cataloging of "standard" or "well known in the industry" methods to be used as case studies and instruction
There is nothing you can "flip" wich will cause you to go "YEAH THAT'S IT". Things like that don't exist. Stop looking for them and concentrate on listening the mix and someday you can really get there. To put it somewhat bluntly, your mixes do not suck because you're missing the knowledge of some "tricks" or "standard procedures". Your mix suck because you can't mix and you can't hear the problems within your mixes. I assure you that even if you had all U87s and all expensive pre-amps and bells and whistles your mixes would still suck. Also, even if you knew how every hit record in the world was mixed and recorded, knew all the "tricks" used in those mixes, knew all the FX settings and all the mics and tehcniques used, it wouldn't make any difference - your mixes would still suck. And I can also assure you that an experienced mixing engineer could do awesome job with that equipment without pulling off any "mixing acrobatics" - just basic compressor, EQ and some reverb and possible other FX. He'd listen and tweak, he'd use his ears and rely on them. He would never think of various "curves" or "applications" or "tricks", he'd just listen and mix it with the basic tools he had.

Last edited by Kainz; 12-14-2009 at 08:17 AM.
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Old 12-14-2009, 09:54 AM   #1080
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Point being, can't say how much money I blew on Marshalls, Fenders, Ampegs, stomps, etc. looking for "the sound".
As a fellow guitar player never forget that 99.5% of "the sound" is in thy hands. Thats not a fancy one-liner, it's the literal truth. Don't let yourself get hung up on equipment, tricks etc, as that is not where the magic is, never has been; whether it be mixing or playing.

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