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I think I've nailed the reason why I prefer the analog medium
Old 13th July 2007
  #1
Gear Head
 

I think I've nailed the reason why I prefer the analog medium

I've come to the realization of the number one reason why I prefer the analog medium of recording. I find that when mixing digitally I am constantly struggling to not go in the red. I know leaving proper headroom is the key, but somehow, somewhere, something is always ****in' in the red and near clipping. It's like a bully who you kicked in the nuts when he tried to take your lunch money and he is now threatening to kill you, causing you to
look over your shoulder at all times. I don't like mixing like that! I think I'm just going to start putting a limiter on every single track.

I feel discouraged sometimes because my mixes aren't ultra modern. I suspect this is because even though I can only record and mix digitally, I treat my DAW like it's a tape machine.

To the people who respect my opinion, thanks for reading my post. I know by posting this I am tossing myself into oblivion, and though I will get owned by just about everyone, that's alright.
Old 13th July 2007
  #2
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f you're recording at 24 bit you have plenty of headroom. Put a trim plugin on each track and lower it by 10 db or so--then stop worrying.

-R
Old 13th July 2007
  #3
Quote:
Originally Posted by RKrizman View Post
f you're recording at 24 bit you have plenty of headroom. Put a trim plugin on each track and lower it by 10 db or so--then stop worrying.

-R
Or just insert VU 's on all your tracks and use those for your meters instead of the peak meters that DAW's normally come with.

I've said it a million times if DAW companies substituted their peak meters for VU meters alot of these "issues" people have with the sound of DAW's would disappear.
Old 13th July 2007
  #4
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davet's Avatar
 

Quote:
Originally Posted by thethrillfactor View Post
Or just insert VU 's on all your tracks and use those for your meters instead of the peak meters that DAW's normally come with.

.
I use Sonar and it had meters on every channel. They appear to respond like VU meters. Is there some difference between a ladder type meter and VU meter? If there's a plugin VU meter that's better than the stock stuff, can you recommend one? TIA,
DaveT
Old 13th July 2007
  #5
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PSPaudioware.com - audio processors and effects plug-ins in VST, DirectX, MAS, AudioUnit, RTAS format for Mac and PC have a free VU meter. But meters take valuble resources, especially if they are calculating average values over a period of time, and displaying a fancy gui.

I believe DAW meters, and the whole way that 50% of the graphical interface is dedicated to the top 6dB where you should never be unless mastering, is responsible for the sick DAW syndrome. IMO, we just have to learn to work around the problem of peak meters.

Orange is the new red.

I know that most of us are trying to get our DAWs to sound as 'analog' as possible. That 'A' word is so overused, it's hard to know exactly what is meant by it. But imo, there are just a few qualities that can be analysed and/or emulated fairly successfully if you know what you want to achieve:

1 - saturation/distortion - there are so many flavours, but basically it's about adding harmonics by altering the waveshape. Tape, tubes, transformers - they all bend the sound differently, but you can analyse a sinewave and watch the harmonics that they add to the sound. I have various types of saturation plugins that can add harmonics, and some even let you adjust each harmonic seperately. It's possible to get very close to the sound of analog distortion. If you are lucky enough to have an analog device that gives you exactly the harmonics you want, that will probably sound best. But sometimes a versitile plugin can get you closer to your ideal sound. I recommend collect as many saturation effects as you can have, especially the free ones, because you just never know when something will be just perfect.

2 - EQ - sometimes the analog gear just isn't ruler flat. So comparing with ruler flat digital sound isn't comparing apples with apples. If you want a particular eq sound, then use an eq. Agreed, that analog & digital eq's sound different - but there is so much choice.

3 - phase shifts - I think a huge part of the 'analog' sound is they way that capacitors and inductors don't react with equal speed across the frequency range. Neither do digital filters - unless you specifically choose a linear phase filter to achieve something that doesn't happen in the real world. But there are plenty of digital phase shifting tricks that can be used to recreate analog effects. I don't see much mention about the use of All Pass filters - but they can be useful in making digital behave more like analog.

4 - compression/limiting - obviously digital has far great dynamic range than analog. Obviously compression and limiting are necessary - but for some reason this seems to be the most abused area. Many legendary analog engineers are famous for being anti-compression. BUT - that has to taken in the context of all the compression they were getting simply by using tape and analog hardware in the first place. And often that engineer was only talking about his own contribution to the process. The tracking engineer, the mixing engineer and the mastering engineer all could apply compression, and just because one of them was anti-compression didn't mean that the other didn't apply a lot.

So if you are trying to get analog sounds out of a DAW, it's a given that you will need compression and limiting. It's just a question of where and how much.

5 - noise - analog creates noise. As much as I hate noise, and appreciate the low noise floor of digital, I think there is something soothing about a certain amount of noise. I can't bring myself to actually add noise (apart from dither noise) but this is something that's starting to bother me. I use various plugins that use convolution technology - and I sometimes hear a lot of noise in the processing that gives an analog flavour - but it's very obvious when the sample dies away, and the noise is gone.

I think it's worth exploring the effects of different types of noise at different levels. At least digital gives us total control over this - i've fought noise all my recording life, and now i'm wondering if I should be adding some ...
Old 14th July 2007
  #6
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AllAboutTone's Avatar
 

Quote:
Originally Posted by RKrizman View Post
f you're recording at 24 bit you have plenty of headroom.

-R
Old 14th July 2007
  #7
Somehow, with analog the challenge is to get UP to the hottest level you can without become distorted, and with digital you have to mix DOWN constantly to keep the overs at arms length. Analogness has trained you to strive and constantly pump everything, whereas digitalness wants you to keep some "open space" at the top so things have air to breathe.
Old 14th July 2007
  #8
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proxy's Avatar
 

Quote:
Originally Posted by thethrillfactor View Post
I've said it a million times if DAW companies substituted their peak meters for VU meters alot of these "issues" people have with the sound of DAW's would disappear.
Sweet merciful crap I wish they'd do that. Built right into the GUI.
Old 14th July 2007
  #9
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Tone Obsessed's Avatar
 

For this very reason, I LOVE the SPL Transient Designers. Can't live without them now. Combined with the Portico 5042 True Tape & Crane Song Phoenix, I'm as close to "Analog" as I have ever been with a DAW.

Have you tried any of these - especially the SPL TD?

IMO, all the "hot" level and "character" you need when tracking by pushing the pre's to the edge allows you to be at UG at mix time with analog type results ( especially true mixing OTB ). No clipping gremlins...

TJ
Old 14th July 2007
  #10
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AllAboutTone's Avatar
 

Quote:
Originally Posted by joelpatterson View Post
with digital you have to mix DOWN constantly to keep the overs at arms length.
hmmmm, i do not have this issue, stand alone 24 bit, the levels NEVER change.
sounds like a sofware, plugin issue.....seems like i went down that road one time when i recorded something on a mbox and adding compression etc with all the rest of the crap that was on there and had same ole issues...tutt
Old 14th July 2007
  #11
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u b k's Avatar
 

make your life easy, and get the best sound out of the daw: insert a trim at the top of every channel strip, and pull it so far back that your meters don't even light up.

when you insert a plugin comp, pull the threshold all the way down and feed it enough input to get the amount of compression you want.

when you go from plug to plug, make sure the level is still so far down the channel meters don't light up.

then, at the output, use another trim to pull the gain back up to the level you want coming out of your d/a.

i have found that running the daw absurdly conservative gives the most open, uncluttered sound possible. it is my opinion that all the notions out there about digital having massive headroom is correct from a mathematical standpoint, but completely the opposite of helpful when it comes to how things sound.


gregoire
del
ubk
.
Old 14th July 2007
  #12
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Dave Peck's Avatar
 

Quote:
Originally Posted by davet View Post
I use Sonar and it had meters on every channel. They appear to respond like VU meters. Is there some difference between a ladder type meter and VU meter? If there's a plugin VU meter that's better than the stock stuff, can you recommend one? TIA,
DaveT


In Sonar, you can set the meters to show peak, RMS, or peak+RMS.

DP
Old 14th July 2007
  #13
Gear Head
 

Thank you all very much for providing your insight. Ubk, what exactly is your definition of "conservative"? Are you saying you exaggerate the headroom you leave during mixing?

Kiwiburger - I appreciate you taking the time to write such a thorough response!

I have a PT LE system and have been recording in 24bit/88.2khz. My problem is that faders creep up on me. I've read gain staging tips such as "the faders feeding a subgroup should be lower than the subgroup fader" and do my best to follow them, but I always feel the urge to crank that snare or kick fader. If I am getting close to clipping the master fader, I tend to lower that instead of the possible problem fader(s) because I feel like I can never make up that volume and retain the same balance by means of subtraction. Does that make sense?

Though my experience recording to tape is limited, I feel like it is more forgiving somehow. I have heard unpleasant tape distortion too. I think I need to get some semi-cheap outboard gear to change up my sonic limited sonic palette.
Old 14th July 2007
  #14
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BradM's Avatar
I think one important thing that nobody has mentioned yet is that in addition to trimming your tracks to give yourself headroom you need to calibrate your monitors so that you are listening at such a level that you won't require you to push your tracks into the red. I would recommend reading up on the K-system developed by Bob Katz. Check out his book called "Mastering Audio". Once you implement the K-system in your control room then headroom will be built into your gain staging and you won't need to rely upon the meters as much.

Brad
Old 14th July 2007
  #15
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turk sanchez's Avatar
If in PT try selecting the "all" group and pull all the faders down a bit when you start clipping.

It's no big deal.
Old 14th July 2007
  #16
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allstar's Avatar
 

Quote:
Originally Posted by waffle waitress View Post
If in PT try selecting the "all" group and pull all the faders down a bit when you start clipping.

It's no big deal.
Or just lower a master fader, if that's where your clipping is happening.
Old 14th July 2007
  #17
Gear Head
 

Quote:
Originally Posted by waffle waitress View Post
If in PT try selecting the "all" group and pull all the faders down a bit when you start clipping.

It's no big deal.
the problem i always run into with that is if i have one of my tracks automated. i pull them all down, press play, and watch that fader go back to where it was. i then just send that channel to it's own aux, and pull the aux down that much, but by the time it gets to that point, and having to do that a good 3-4 times in a session, it becomes a pain in the ass and i wish there was a better way around it.
Old 14th July 2007
  #18
Gear Head
 

Quote:
Originally Posted by FattMusiek View Post
I've come to the realization of the number one reason why I prefer the analog medium of recording. I find that when mixing digitally I am constantly struggling to not go in the red. I know leaving proper headroom is the key, but somehow, somewhere, something is always ****in' in the red and near clipping. It's like a bully who you kicked in the nuts when he tried to take your lunch money and he is now threatening to kill you, causing you to
look over your shoulder at all times. I don't like mixing like that! I think I'm just going to start putting a limiter on every single track.

I feel discouraged sometimes because my mixes aren't ultra modern. I suspect this is because even though I can only record and mix digitally, I treat my DAW like it's a tape machine.

To the people who respect my opinion, thanks for reading my post. I know by posting this I am tossing myself into oblivion, and though I will get owned by just about everyone, that's alright.
i'll be honest, when i have one snare channel, that has one or two clips in the song, i just throw a limiter at -.01 on the last insert, and forget about it. to me, it's not audible to hear those few clips, and all the limiter for me is doing is keeping that damn red light off.

oh, and to your main point, i feel the same way. mixing in the DAW is a pain, because you spend 70% of the time battling clipping and your levels, and when you're mixing OTB, you're just mixing. ITB really kills my creative flow.

I WILL be trying the trim plugin next time though...
Old 14th July 2007
  #19
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mixerguy's Avatar
it is all about gain structure. plain and simple.
Old 14th July 2007
  #20
Most analog consoles sounds good when in red (and often also sounds good when not in the red), daws just never sounds like anything, except for what you feed it with, and then some jitter, truncation, fan noise and stuff.
Old 14th July 2007
  #21
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Zep Dude's Avatar
 

I think we all have this problem to one extent or another, but the solution is to change your methodology when working ITB.

#1 With the first thing you bring up in the mix (usually drums), hit the master bus at a lower level than you have been hitting it (reduce by 6db or even 10db). Now as you bring up everything else you won't be clipping.

#2 If you are still clipping it is most likely only a few times in the song where many things hit together. Figure out the offending element(s) and duck the transient in the automation, or gain reduce them with the audiosuite "gain" plugin by a few db.

#3 If you're still clipping often, put a limiter with fairly fast attack on the offending track. Set the threshold to only trigger on the loudest hits so it's invisible most of the time and only kicks in when necessary.
Old 14th July 2007
  #22
Quote:
Originally Posted by tommy lee View Post
Most analog consoles sounds good when in red (and often also sounds good when not in the red), daws just never sounds like anything, except for what you feed it with, and then some jitter, truncation, fan noise and stuff.
Which analog consoles are these?

On newer SSL's when you are in the red it sounds like SH*t!!!

On the older SSL's it sounds like a big headache.

Older Neve's have a lot of headroom but when you are in the red there is no 2 track recorder analog or digital in the world that won't distort when you try to record this signal.

On other consoles you may not notice it as much on one channel but you add the distortion together on different channels and your mix will sound small with no bass.

Mixerguy is right.

If you can't get the gain structure right in analog right there is no way you will get it right in digital and vice versa. Remember the ideal setting on a fader is zero. They were designed to blend the signals not change gain structure. That's what the trims on the consoles are for. Its engineering 101.
Old 14th July 2007
  #23
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u b k's Avatar
 

Quote:
Originally Posted by FattMusiek View Post
...what exactly is your definition of "conservative"? Are you saying you exaggerate the headroom you leave during mixing?

when i say conservative i mean more conservative than you ever thought was wise. i'm not exaggerating when i say i don't usually see any life on the channel strip's meter; if they light up, it's not by much.

when i'm coming into, say, a plugin compressor, my threshold is usually down around -40db to give me 2 or 4db gain reduction... that's how little signal i'm feeding it. i am equally parsimonious with what i feed every plug, every chain. then there's another trim on each physical output's fader doing 15-25db makeup gain in order to give me sufficient level to hit the outboard.

ime, running the math so conservatively preserves tone, transients and depth, and maximizes punch and clarity if i do any submixing. when levels start to get hot, all of the above begin to diminish markedly, and if the mixer is lit up light an analog desk then everything folds. i don't know if others have this experience, but i'm clear on what i hear and what the remedy is for me.

enjoy!


gregoire
del
ubk
.
Old 14th July 2007
  #24
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Computers suck. There, I said it.
Old 14th July 2007
  #25
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peeder's Avatar
 

Quote:
Originally Posted by Kiwiburger View Post
5 - noise - analog creates noise. As much as I hate noise, and appreciate the low noise floor of digital, I think there is something soothing about a certain amount of noise. I can't bring myself to actually add noise (apart from dither noise) but this is something that's starting to bother me. I use various plugins that use convolution technology - and I sometimes hear a lot of noise in the processing that gives an analog flavour - but it's very obvious when the sample dies away, and the noise is gone.

I think it's worth exploring the effects of different types of noise at different levels. At least digital gives us total control over this - i've fought noise all my recording life, and now i'm wondering if I should be adding some ...
Noise masks low-level harmonics and other forms of distortion that can be very distracting, much more distracting than the noise itself is. I think this is probably the major difference in the list.

Consider keying a gate on a noise generator to a signal and blending noise in until your signal sounds best to you. Sure it might be best of all to not have the distortions in the first place, but if you have them, you can mask them this way without people noticing.
Old 15th July 2007
  #26
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Quote:
when i'm coming into, say, a plugin compressor, my threshold is usually down around -40db to give me 2 or 4db gain reduction... that's how little signal i'm feeding it.
Thanks UBK - makes me feel like i'm not so crazy.

I find software compressors need far more headroom than you would think to avoid unwelcome distortion. The clip indicators are way too late imo. And I usually find the autogain features need to be turned off because they do too much damage.

I think DAWs get blamed for a lot of damage done in the analog realm too. For example - because of the skewed view that peak meters and waveform displays offer (focusing on the top 6dB), when tracking it's too easy to feel that we need to see some signal in that area. But in order to get signal that hot, our preamps have to be cranked hotter than they need to be.
Old 15th July 2007
  #27
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pkautzsch's Avatar
 

Not as conservative as UBK, but I usually go into the DAW peaking at about -9 dBFS for each channel. Try to think -9 dBFS is 0 VU. Usually no problems when you're not misusing compressors' makeup gain.
Old 15th July 2007
  #28
Quote:
Originally Posted by FattMusiek View Post
I've come to the realization of the number one reason why I prefer the analog medium of recording. I find that when mixing digitally I am constantly struggling to not go in the red. I know leaving proper headroom is the key, but somehow, somewhere, something is always ****in' in the red and near clipping. It's like a bully who you kicked in the nuts when he tried to take your lunch money and he is now threatening to kill you, causing you to
look over your shoulder at all times. I don't like mixing like that! I think I'm just going to start putting a limiter on every single track.

I feel discouraged sometimes because my mixes aren't ultra modern. I suspect this is because even though I can only record and mix digitally, I treat my DAW like it's a tape machine.

To the people who respect my opinion, thanks for reading my post. I know by posting this I am tossing myself into oblivion, and though I will get owned by just about everyone, that's alright.

Nah... I think I felt a lot like that through much of the early 90s working with 16 bit i/o and unforgiving systems.

I have to say that, while I still keep an eye on proper gain-staging for channels in my DAW that have plugs, working in a modern DAW with FP math takes a lot of the onerous weight off. I simply track more conservatively than I felt driven to by the exigencies of 16 bit AD/DA -- and feel more comfortable doing so. I really could hear the effect of tracking too low on my old 16 bit rigs. But -- of course -- obviously, you've still got gain considerations -- you can't get sloppy, you've still got an analog stage to optimally gain set. And inside the box, you may not have to worry about overs directly in your floating point DAW -- but you still have to worry about plug ins. And, IIRC, at least one very popular host-based DAW, PT LE, which does have a 32 bit FP engine, has a 24 bit fixed path to plug ins -- so there, you absolutely HAVE to watch your internal gain staging where plugs are involved. (Hope I got that right. Feel free to correct me, please.)

And, heck fire, maybe the rest of y'all got to work on nice, quiet high end analog gear... but I worked in a gritty real world of noisy, ill-maintained consoles, old or small format decks, and tinky outboard gear... I ALWAYS had to keep my levels up in order to fly above the gunk -- but then watch out for the artificial ceiling of nutty NR schemes like dbx NR... [which I often turned off for drums and basses, of course, in order to actually get some traction out of the tape medium] Anyhow... it's all tradeoffs... I'm definitely liking digital in the 21st century a lot better...
Old 15th July 2007
  #29
Quote:
Originally Posted by Dave Peck View Post
In Sonar, you can set the meters to show peak, RMS, or peak+RMS.

DP
True.

You can also set the meter rise and fall times, peak hold times (on the Options / Audio Metering Options dialog), as well as various scaling options, as well as segmented vs continuous metering. But no VU weighting, per se. Still, I find the RMS+Peak pretty useful, I have to say.



Coming up in the era of mechanical meters, when instantaneous, fully responsive metering was impossible and peaks had to be specially handled in order not to overload meters, I understand the thinking that went into VU meters and the smaller delays/loading that went into PPMs. (Bell developed VUs for standardizing levels in the telephone system. The BBC developed peak programme metering for radio work. Both were in the mid-late 30s. Even the PPM of the day had about a 10 ms response time, in order to protect the meter.)

When true peak metering with virtually instantaneous response times became widely available, I saw the immediate beauty of it. Others had problems adjusting... since tape saturation was considered desirable by the time I was working in studios, the "superiority" of the intentionally sluggish response times of VU loaded metering was already becoming a widely accepted notion. Peak metering "confused" fader pushers used to VU response curves with their sluggish response to transients.

But, of course, while quick transients were "absorbed" in tape saturation, digital is far less forgiving. For THAT reason, I think soft and hardware manufacturers assume that we recordists would prefer to know just what's going on with our transient levels -- since they can and will crash out a digital system.

You know, if you like flynig blind, that's cool. I don't. Give me peak metering.
Old 15th July 2007
  #30
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Marjan's Avatar
 

Quote:
Originally Posted by Megalomaniac View Post
Quote:
Originally Posted by waffle waitress

If in PT try selecting the "all" group and pull all the faders down a bit when you start clipping.

It's no big deal.
the problem i always run into with that is if i have one of my tracks automated. i pull them all down, press play, and watch that fader go back to where it was. i then just send that channel to it's own aux, and pull the aux down that much, but by the time it gets to that point, and having to do that a good 3-4 times in a session, it becomes a pain in the ass and i wish there was a better way around it.
You don't need to add Aux's.
If you select the "All" group and then lower the fader volumes in the Edit Window, all the volumes will come down in relation to each other, including automated assignments. And you can of course do it on individual tracks as well.
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