I don't completely buy into the obsolescence theory. You all have upgraded your analog gear over time. Yes the mic pres and outpboard compressors are pretty much timeless but the guys who bought 4 track 1 inch eventually upgraded to 8 track then 16 track then 2 inch then 24 track etc. Those tape machines each cost the equivalent of 2 or 3 full blown TDM systems.

I just happen to think hardware compressors are outdated. They are not "required" in order to get level, and I think there are enough software compressor choices that sound very good, Waves RenComp, Sony Dynamics, McDSP, UA, that I just throw the compressor on a track and record with the plugin in place ready to go. Since the talent is monitoring the TDM output they hear the compressed sound as they track. This has all the advantage of printing compression to tape with a couple of convenences:

1. I don't have to have the singer suffer through compressor trials and selection. I can just put up something appropriate and move on.

2. I can change it later if I like or let the mix engineer use whatever he or she likes.

3. I don't have to waste i/o on outboard stuff.

4. I can keep my signal path clean and take advantage of all 24 bits worth of resolution (unlike using a compressor to get signal level which amplifies the noise level and reduces my detail resolution just to make it louder).

Steve

Originally posted by sdevino:

Originally posted by bunny:
Personally, I don't see how you're going to safely get anywhere near "Digital 0" without a little compression going in. I use a lot of analog gear and record to an MX-2424.

There is no reason to try to get to digital zero in the first place. You are better off just running the analog input chain at whatever level it sounds best at. This will usually be somewhere around 0VU maybe a little higher, maybe a little lower on some pieces. A well calibrated studio sets 0 VU to about -14 to -20dBFS. Its good to have head room. Anyone who thinks you "lose bits" when you track below digital zero simply doesn't understand digital audio.

Steve So there's no loss of sonic detail or "three-dimensionality"?

I have some replies to make regarding Steves last comments..

Originally posted by sdevino:
I don't completely buy into the obsolescence theory. You all have upgraded your analog gear over time. Yes the mic pres and outpboard compressors are pretty much timeless but the guys who bought 4 track 1 inch eventually upgraded to 8 track then 16 track then 2 inch then 24 track etc. Those tape machines each cost the equivalent of 2 or 3 full blown TDM systems.

A full blown TDM system the last time I checked, is around $20,000. It is obsolete in 16 months (Digi-planed obsolecence-designs release cycle) I know of poeple that are still useing their 1" 8 tracks after 20 plus years.. I can't think of one example of a 16 bit TDM system still in use.. what has it been, 5 or 7 years since Digidesign was doing 16 bits?

I just happen to think hardware compressors are outdated. They are not "required" in order to get level, and I think there are enough software compressor choices that sound very good, Waves RenComp, Sony Dynamics, McDSP, UA, that I just throw the compressor on a track and record with the plugin in place ready to go. Since the talent is monitoring the TDM output they hear the compressed sound as they track. This has all the advantage of printing compression to tape with a couple of convenences:

I myself find compressing to disk almost more important than compressing to tape. Tape is more forgiving. Digital mediums do not tolerate overs at all.

1. I don't have to have the singer suffer through compressor trials and selection. I can just put up something appropriate and move on.

Neither do I. I use an EL OP at a gain reduction of 2 to 3 dBs.. This takes seconds to set up.

2. I can change it later if I like or let the mix engineer use whatever he or she likes.

I never find the need to change what I have done on the way in. In some instances, I may add a little more compression at mix. In that scenareo, I can use the stock plugs that came with my recording software and the sound of the EL OP, LA4 or the 1178 is already imparted to the track. It's a matter of having enough expierence and confidence in yourself to make an initial desision and then live with it.

3. I don't have to waste i/o on outboard stuff.

Neither do I. I just compress a bit on the way into the box.. At mix, I use plugs.

4. I can keep my signal path clean and take advantage of all 24 bits worth of resolution (unlike using a compressor to get signal level which amplifies the noise level and reduces my detail resolution just to make it louder).

I actually feel I am getting into the box with more bits, or aleast more level, being able to comfortably track at -6dB, digital scale, instead of -12dB to allow headroom for peaks..

No. In fact quie the opposite. The limit in detail is the noise floor. A VERY good analog input might have 110dB signal/noise under ideal conditions. The 24 bit converter is capable of 144 dB so the limitis already in the analog domain. If you add an extra 20 to 14 dB of gain using a compressor inorder to get the input level up to 0dBFS you are also making the noise floor higher. So instead of 110 dB you have 90 dB s/n at the input.

Looking at it another way, a 24 bit converter can resolve changes in the input signal of 1/16777216th of full scale, no matter what the input level is. The noise floor of the converter's analog front end limits this to about 1/1000000th of full scale. Adding 12 dB of gain using a compressor limits it to about 1/250000th. So you lost 12 dB worth of detail to the noise floor.

These numbers are for illustration only since you can resolve a sound below the noise floor but it should still make the point.

I have found my mixes sound fuller, warmer and more detailed when I leave some headroom while tracking. Typically I try to track at around -18 to -14dBFS depending on what sounds better.

Steve,
By having to boost the volume of every track at mix you are making the DAW perform more calculations and thereby (in theory) making the sound more ragged. It is a well accepted convention that the sound is at it's purest with the faders set at 0dB.. I personally don't have any problem with noise floor. Good mics and front end gear should not exhibit these propensities at all.

Originally posted by Kurt Foster:
Steve,
By having to boost the volume of every track at mix you are making the DAW perform more calculations and thereby (in theory) making the sound more ragged. It is a well accepted convention that the sound is at it's purest with the faders set at 0dB.. I personally don't have any problem with noise floor. Good mics and front end gear should not exhibit these propensities at all.

This has been proven to be untrue (see the 3D DAW CD).
Noise floor is what will limit the detail. My point is that if you raise the analog noise floor, you are reducing the ability of the converter to capture detail because it is masked by the noise. There are some very very long threads where this has been born out. This is simple gain stage mangement and is as valid in the digital domain as it is in the analog domain.

There is no reason to avoid using the computer to do some processing. this is another tired old wive's tale. 48 bit fixed point math in a well designed plugin sounds great. the raggedness you refer to is well below -144 dB in a 48 bit system.

when I do a recording I track around -14 to -18dBFS. When I mix my channel faders are typically somewhere between -10 and +3 dB. If I need to make up gain it will come from a Waves C4 or L2 or Rec Comp or Sony Oxford. These mixes sound large, full, detailed, 3D and warm.

When I track too hot or run levels close to 0dBFS I start losing detail and the mix starts to get a boxed in screechy digital tone to it.

My ears say it sounds better, and science says it should sound better.

Originally posted by Kurt Foster:
I have some replies to make regarding Steves last comments..

Originally posted by sdevino:
I don't completely buy into the obsolescence theory. You all have upgraded your analog gear over time. Yes the mic pres and outpboard compressors are pretty much timeless but the guys who bought 4 track 1 inch eventually upgraded to 8 track then 16 track then 2 inch then 24 track etc. Those tape machines each cost the equivalent of 2 or 3 full blown TDM systems.

A full blown TDM system the last time I checked, is around $20,000. It is obsolete in 16 months (Digi-planed obsolecence-designs release cycle) I know of poeple that are still useing their 1" 8 tracks after 20 plus years.. I can't think of one example of a 16 bit TDM system still in use.. what has it been, 5 or 7 years since Digidesign was doing 16 bits?

This is a choice. There are in fact many people still using 16 bit systems for post etc. The point is 24 bit 44.1ksps is as much as anyone needs. Upgrading is a choice. Just like keeping your old 8 track 1 inch was a choice when 2" 16 track came out.

I myself find compressing to disk almost more important than compressing to tape. Tape is more forgiving. Digital mediums do not tolerate overs at all.

And I solve this problem by tracking lower or more appropriately at a level where my analog input chain is in its sweetspot (which is not usually at +10dbVU). BTW soft clip features on most modern converter systems are very forgiving.

Neither do I. I use an EL OP at a gain reduction of 2 to 3 dBs.. This takes seconds to set up.

2. I can change it later if I like or let the mix engineer use whatever he or she likes.

I never find the need to change what I have done on the way in. In some instances, I may add a little more compression at mix. In that scenareo, I can use the stock plugs that came with my recording software and the sound of the EL OP, LA4 or the 1178 is already imparted to the track. It's a matter of having enough expierence and confidence in yourself to make an initial desision and then live with it.

You just reinforced my point, You are really using the compressor to modify the overall sound, like an effect. This is personal style and choice. I just happen to choose other tools to get the sound I want. I am not usually thinking in terms of getting the sound of an ELOP or LA2. I just don't think that way. I just go for a sound that fits the song, whatever that myight be. Again a matter of personal working style.

I actually feel I am getting into the box with more bits, or aleast more level, being able to comfortably track at -6dB, digital scale, instead of -12dB to allow headroom for peaks.. The I am "getting into the box with more bits" is the part where you are making a mistake. Most people just seem to think of this backwards. The detail is all in the LSBs. If you have a -20 DBFS signal going into a 24 bit and a 16 bit converter at the same time:
- The 16 bit converter will detect changes 76 dB smaller than the input signal.
- the 24 bit converter will detect changes 124 dB smaller than the input signal.

The real limit however will be the noise floor. Which on your EL OP or LA2A is probably about 90 dB below 0VU at best. If you then add 10dB of gain you raised the noise floor of yuor ELOP to -80dB. So now your 24 bit converter will only see levels 70 dB (signal is -10dBFS and noise floor has been raised to -80dBFS).

OK one more time:
- the whole world is free to use a compressor to get a certain sound while tracking. this is very cool.
- on the other hand using the compressor to get hotter level to HD is often counter productive.

Steve,
Agreed.. I personally couldn't see myself using a high end plug in. I like my outboard and it has been bought and paid for.. it also has earned it's keep. If I were just starting out, I might consider the approach you advocate, but I still see the issue of having to buy a new set of plugs each time Digidesign or Mac comes out with the next version of hadrdware, software or OS.. as a deal breaker. This to me is a major deterent. Add to that, the issue of obsolecence.. old plugs are basically worthless and my EL OP, 1178 LA4's and Dynamite are worth more today than when I purchased them.. But as you point out, we all have our different slants on things and one of the wonderful things about audio is there are many paths from here to there. If a person gets the results they are seeking, they have done it "right"... Kurt :tu:

Very cool Kurt. I have had to pay on average about $500 per upgrade to get all my plugins up to date. which is a lot less money than I spend on gas taking my teenagers to play and sports practice. And my plugins almost never cause me to lose sleep at night.

Anyway given that this is my business I find this amount to be very acceptable. If I were doing this for fun I would strongly recommend peole get what they need and keep it and resist the upgrade urge.

Steve, I can see your point.. When I was in business, I too, would rather go buy something at the end of the year rather than give the money to Uncle Sam.. running a commercial operation requires some expense.. You are fortunate indeed to be in that position. Many guys are scrapeing by trying to keep the doors open and competing with the home recording revolution.. Good for you. I guess you are not without power today? (obviously) Kurt

I have not felt the need to go PT HD yet. If I had lots of money, I would have certainly done it already, but not the case. People bash, complain, we get pissed off with long bug lists and so, but Pt is here to stay, at least for a few years more. If ya want five star midi, go Logic, DP. Anyway, there are several tools out there, nice music is the goal, no matter if it is being done with $$$ analog hardware ( Manley, Crane, avalon)or a small to medium PT Mix/HD.

I wish I had a ton of vintage (or even modern) hardware. Maybe someday I will.

But isn't the argument that if you have mostly computer hardware and software you will have to start from scratch every few years because of obsolescence a little misleading?

I was under the impression that most DAW software and Plug-in manufacturers had upgrade paths and pricing. So that the $20,000 system wouldn't require an additional $20,000 every couple of years. Maybe only $5000?

That's not chicken-feed, I'll agree. But not the same as total obsolescence either. But maybe some systems don't do this?

Digidesign just recently released the HD96k version.. now they are scrambeling to upgrade to accomodate the new Mac G5.. it is a constant state of upgrade. As I said 7 year old 16 bit PT systems are obsolete.. you can't use the plugs from a 16 bit system on a 24/48k bit or a 24/96k PT rig ...

I have vintage and legacy as well as modern versions of hardware compressors and eq's that will be useable when your grandchildren are in college..

Yes upgrade paths are available as Steve mentioned for $500 per plug in.. I don't know about you but once I have spent my money on a product I don't want to have to put more money out to be able to continue to use it. Good money after bad, type of thing...

Heh, heh, funny that the best way to stay ahead of the digital curve is to go analog. David

The upgrade thing is a bastard, although I know someone with PT3 rig and way old mac. No one taht uses his studio knows its not the latest PT rig.
I do not think they care anyway.
A friend still uses his atari/notator for midi.

The thing people forget is you do not have to update if it is working for you...

In "pro" studios, it *should* be about the sound, not the money. If you feel that you can get the best sound using softwareplug-ins and feel that the cost is justified, then go for it!

If you feel that you get better sound using analog gear going in, then you are obviously making the right choice by doing so.

The decisions should come down to the "sound" not the economics...that is *if* I am reading the tons of other threads around here about "room size", "jitter" and who knows what else...

If you make an argument in one thread that cost is not the real issue, it is the quality, then you need to be consistent and make the same argument here...

In my opinion, in any commercial endevour, cost *has* to be considered, no 2 ways about it! It is always a balancing act between cost and quality if you want to run a profitable business...it is just simple economics!

Personally, I am not convinced thatplug-inscan impart the same feel that their analog counterparts do. I spent too many years in studios with tons of great analog gear (which I wish I had!) and made some really great recordings! For me, I am not able to afford the really great vintage analog stuff, so I have to compromise and use theplug-ins..I do have faith that someday, we will be at the point where the difference will be negligable...and probably sooner than we think!

*My* biggest gripe about the digital world is not the quality of the digital sound, but rather the endless takes, re-takes, punch-ins, quantizing, pitch correcting...etc...it has taken the life away from the music...

I used to be pretty pissed off by the hype and dis-service by the plug makers claims and the clients that would insist or expect that I have them. Since I usually had the hardware versions of most of what they asked, they would shut up when they heard the real thing. After thinking about it some, I've come to endorse the plugs as well as the clean look of min amount of rack gear as best for my clients and in my best inerest. Not because it sounds as good or better, but because I want them and everybody else to dump their rack gear hardware for cheap, and then I can buy it up. So now I hope plugs are all the rage for the home wanna-be wankers, newbies and everybody inbetween. Tell all your friends, clients, everybody you know to jump on the plugs bandwagon and dump the hardware and then us true gearslutz can aquire the stuff nice and cheap.

Sure, this will ruin the re-sale market for hardware. But then I never bought the rack hardware ever thinking I was going to sell it, yet alone try to sell it and make profit or close to money back. The boyz with the most toyz that make a great tone, wins!

The irony of it is the outboard gear has gotten more expensive. As a studio owner I see it as overhead and repair time.

My per month DAW expenses continue to go down over time even though the capability is increasing.

I also find my clients very greatful for the speed at whcih we can get things done on the DAW and the cost of recording media vs analog.

Most of my clients have limited budgets so doing excess takes, punchins and endless auto tune are not really options (time wise).

In big production projects we do spend the time to get things right, fix method is determined by what makes the most sense. If it easier for the bass player to punch in a fix then we do that. If it is easier to fix it with an edit then we will do that instead. Its all about options.

The biggest advantage I see in digital recording is the ability to capture acoustic nuance with great detail This means we do not use as much close micing and do lots of stereo tracking. I think this is a critical advantage of digital recording. In my opinion close micing never really sounds like the original instrument. Instead we have learned to expect certain instruments to sound a certain way on recordings (i.e a snare recorded with a SM57). Tape helps maks some of the scratchy detail of close micing while adding a high end hiss which is pleasing and therefor makes this type of recording sound better.

Digital on the other hand doesn't mask or enhance so all you get is the original scratchy detail. On the other hand, if you move a little further back and get some ambiance into the recording and possibly even use a stereo mic setup, digital does a much better job of capturing the details of the instrument's natural sound along with the sound of the room then analog can. This has a couple of advantages:

1. The instruments sound more like they do in the room. So you can get it to the sound you want without having to playback tape just to see what the final will sound like.

2. The ambience in the tracks can (if well thought out during tracking) help place the instruments in the mix in a very 3D way without having to resort to EQ or delays (i.e less processing Kurt!).

The idea is to take advantage of the strengths of digital, which is it plays back exactly what you feed it. SO if you make it sound great before you hit record you can pretty much avoid lots of processing. Analog does not do this. Tape sounds different when played back. Luckily most of us like this sound better and we almost use tape as a processor as well as a recording medium. If you try to use this method with digital you will be very dissapointed.

Originally posted by sdevino:
The irony of it is the outboard gear has gotten more expensive. As a studio owner I see it as overhead and repair time.

I have never had any expense maintaining my LA2, LA3's LA4's and only a small expense once in a while to replace the 6072's in my EL OP.. However my computer has been to the shop 3 times this year..

I'm with AudioGaff... hey Steve, Do you have any hardware outboard you want to sell cheap??? :D

Originally posted by Double Helix:
In "pro" studios, it *should* be about the sound, not the money. If you feel that you can get the best sound using softwareplug-ins and feel that the cost is justified, then go for it!

If you feel that you get better sound using analog gear going in, then you are obviously making the right choice by doing so.

The decisions should come down to the "sound" not the economics...that is *if* I am reading the tons of other threads around here about "room size", "jitter" and who knows what else...

If you make an argument in one thread that cost is not the real issue, it is the quality, then you need to be consistent and make the same argument here...

I don't know if that is directed at me or not but I will say that buying hardware is the more expensive way to go. But IMO it is the better investment. I always encourage folks to bite the bullet and put the big dollars where the investment and quality factor lies.. I don't like the idea that software companies seem to have this plan to milk the customer base repeatedly through planned obsolesence and upgrade-itis ... If you go out and buy a Manley or a UA product, that is probably the last you will ever spend on it (other than tubes once in a while) and you will be able to use it forever, no matter what software or computer you are useing. And if things ever go full circle back to analog (one can hope, can't they?) you will be ready for it.

There is no doubt that hardware is an investment, and software is trash money... I am with you there. I also agree that to my ears, nothing can touch the analog outboard gear... (YET!). There is an initial higher cost associated with the hardware, but the resale value makes it a better buy.

I guess my point was that the pro studios *should* be making decisions based on sound quality alone, not on cost, but unfortunately, they also need to make money, so this is not always possible, especially in this economic environment. Everything is a balancing act.

Go for what your ears tell you...for Kurt, it is analog outboard gear and for Steve, it is plugs. Like you said earlier, these are just different paths to the same end. For me, I am an "amateur" and stuck in the plug-in world for most of my processing...some day however... :)

Digital doesn't sound just like the input source anymore than analog does.
That's hype.
Anyone who cannot tell console out from the playback through the A-D and D-A is in the wrong business.

It can sound good, it can be fine.. but it's NOT "just like" what goes in.
Not yet.

Originally posted by wwittman:
Digital doesn't sound just like the input source anymore than analog does.
That's hype.
Anyone who cannot tell console out from the playback through the A-D and D-A is in the wrong business.

It can sound good, it can be fine.. but it's NOT "just like" what goes in.
Not yet.

OK you need to fill this one in. If I am monitoring through my playback path when I record (which is what I do), then why will it sound different when I play it back from HD without the record button pushed?

Moniotring through playback means the data is being converted and routed through the TDM mixer. plugins and out the speakers the same way as it does on playback.

It will sound the same because it is exactly the same.

What were you talking about????

OK I read your post again. We are talking about 2 different things. I am not saying that digital is making a "perfect" recording.

I am saying that:
1. you can monitor via the digital mix path including plugins etc, so you can hear playback quality while you record (on a TDM system only).

2. Once you hear playback, it will playback the same everytime.

3. When you record to tape, tape playback sounds very different than console out (I do not use a console for digital).

4. when you mix from tape, you pass the tape over the playback heads many times, so you gradually wear the tape down and the quality changes. Good engineers plan for this and record with a little precompensation for tape wear.

Steve,
This may open the door to another debate but Nyquist filters that filter ultra sonic frequencies.. that is probably the main reason.. also bit depths that are not capable of reproducing full dynamic range is another. There is not a recording system yet, that truly puts out exactly what it is fed..

Originally posted by Kurt Foster:
Steve,
This may open the door to another debate but Nyquist filters that filter ultra sonic frequencies.. that is probably the main reason.. also bit depths that are not capable of reproducing full dynamic range is another. There is not a recording system yet, that truly puts out exactly what it is fed..

I wasn't arguing this point. BTW, while I don't think there is enough evidence to argue either way on the importance of ultrasonic frequencies (we'll all learn the truth about this someday I hope), anything over 18 bits has more than enough bit depth to capture the full dynamic range of anything in the analog audio world.

Steve,
I may be mistaken but aren't real world dynamics like 150 to 160 dB?? I know there isn’t a mic capable of reproducing this spl ... but 24 bit is like 110 dB dynamic range.. not arguing, just questioning... Kurt

Originally posted by Kurt Foster:
Steve,
I may be mistaken but aren't real world dynamics like 150 to 160 dB?? I know there isn’t a mic capable of reproducing this spl ... but 24 bit is like 110 dB dynamic range.. not arguing, just questioning... Kurt

Its not exact but if you want to calculate the dynamic range of a digital system add 6 dB for each bit (i.e. each n bit doubles the dynamic resoultion).

Now back to your question:
- Prior to the 1990's the best analog channel available in commercial equipment (other than lab gear) had a dynamic range between 90 and 96 dB (most gear was less than 90).
- The best tape format was probably in the 70s to low 80s without noise reduction.
- 21st century analog gear can be as high as 110dB but is more likely around 90 in actual use.

A case in point, the newest Audio Precision audio test system has an input THD+Noise = -112dB.

16 bit digital has 96 dB of dynamic range (pretty much as good as it ever got in analog).

18 bit is 108 dB about the same as the best contemporary stuff.

20 bit is 120 dB

24 bit is 144 dB.

You get all of this resolution within the digital environment (i.e in the mixer and plugins). But the A/D converter is limited to the analog front end of the converter's dynamic range which will be 90 to 110 dB depending on how it is implemented.

The limit of the analog noise floor is due mainly to the thermal noise caused by electrons flowing through resistive wires and components. It is VERY difficult and requires very talented design engineers to make small (like 2-3 dB improvements) in the analog dynamic range.

Analog and digital have the same dynamic range through the ADC but once you are in the mixer the digital has far more dynamic range (the Pro Tools TDM mix engine has 288 dB of dynamic range).

Here is an interesting article on digital mixer gain management:
http://www.digidesign.com/digizine/techtalk/index.cfm

Steve

Okay....forgive me for an ignorant slut...but if its true(i assume it is) that PT's mix engine is capable of 288db of dynamic range, then why are most recordings we hear today, most of which are probably protooled to some extent,why are these recordings exhibiting a dynamic range of +/- 3db at BEST?

Is this what we're shooting for?

I remember music having really loud and really soft passages. Things that had passion and emotion and strength even in the quietest moments.

someone pleez 'splain dis to me......

It's all about the "Bling Bling". The more "Bling", the better, right? :eek:

Didn't you know? :c:

I hear, enjoy and experience more dynamic range out of all my old 33RPM records than anything I've ever heard released to the public from protools...

Discussing or even trying to achieve wide dynamic range is pointless when we all are guilty of contributing to the popularity of lifeless music by either buying it, or worse yet, are involved with the production of it.

It is the gear manufactures that are desprate for sales and to keep from going out of business that insist on how we all are suffering if we don't hurry and jump on the higher bit and sampling rates while we squeeze the life out of the music to make it louder, then squeeze it down to fit 16-bit where the consumer cares not, so they squeeze it down it to MP3 and then play it on squeezed down computer speakers that are not even capable of the crappy dynamic range of MP3.

My comment on digital recording had to do with the statements that implied that analog simply "masked" or "flattered" close mic'ing or other techniques thereby making them acdceptable ina way that digital did not.
This is not my experience.
For one thing I get a sound I want listening to the desk OUTPUT.
In other words, it sounds the way i want it BEFORE it gets to tape OR digital.
The system that can spit that back exactly would get my vote.
But none does.
Analogue tape changes the sound in a (usually) musical way, but changes it.
Digital recording simply 'misses' bits of the sound (like pouring water over an array of closely spaced glasses.. some falls in the cracks).
But if i am getting the snare drum sound i WANT from a clsoe mic, i am not relying on tape to 'make it work'... rather, i am hoping the chosen format doesn't screw up my already good snare sound.

ps i wouldn't be caught dead using an SM-57!

Originally posted by wwittman:
i wouldn't be caught dead using an SM-57!

How about a Beyer N201??

Sorry gang, but I think that the overuse of your beloved compression is cause for loss of dynamics in music. Pro tools is an outstanding tool for recording classical music which still has plenty of dynamics as far as I know.

wwittman wrote:

Digital recording simply 'misses' bits of the sound (like pouring water over an array of closely spaced glasses.. some falls in the cracks).

You will have explain this one too. Nyquist says this is not true and I have had the opportunity to prove that Nyquist is correct. I can sample a waveform with 2 samples in a given cycle (the sampled values end up looking like a square wave) and then perfectly reconstruct the origina waveform from that. Nothing is missing. Since we sample at >2x 20kHz and the very highest bit of detail we can hear is around 20kHz (as far as we know today), then anydetails that fall between the cracks are made up of frequency components greater than 20kHz (whic leads to Kurt's argument).

The difference with digital is I can monitor AFTER the conversion process, with tape I have to wait for playback to hear what it did.

And while the SM57 is not my favorite mic sometimes it is the right mic for the job.

Steve

sm57?

I have recorded on 16 bit digital ... I have recorded on 24 bit digital and 24/96 digital. I have also recorded on 2" analog both at 30 ips and 15 ips. I find that while all these formats are useable and capable of turning out excellent product, none of them are perfect. I'm sorry Steve, the perfect playback system has not been invented yet. The day it is, everyone in the world will know about it. Believe me it won't be a well kept secret.

Modern day analog and digital systems do not reproduce "real world" dynamics or frequency response and all exhibit some undesirable side effects. Add to the mix that different individuals hearing varies and it is impossible to ascertain what the absolute highest frequency humans are capable of hearing is. All of this doesn't even take into account that there is a possibility that higher super ultra high frequencies may generate a "fold back" effect and excite lower audible frequencies. All the facts are still not in, the data is incomplete.

Part of the art is in making these systems sound like the “real” thing or at the very least, believable. I think that compression is a valuable tool in this process. I prefer to compress a bit at the time of tracking and then perhaps a bit more at mix.. I almost never used compression across the 2-bus when I worked with an analog console but I have to admit I am doing this more now that I am working in DAW. I have no explanation as to why, it just seems to deliver a better product that way.

I hate over compressed music. Even though I am the “compressor pig” of the West Coast, I still like music with dynamics.. Much like my not being able to drink coffee without a spoon of sugar but not being able to drink it with 2 spoons of sugar, compression is a wonderful thing in small amounts.. Kurt

Originally posted by Kurt Foster:
I have recorded on 16 bit digital ... I have recorded on 24 bit digital and 24/96 digital. I have also recorded on 2" analog both at 30 ips and 15 ips. I find that while all these formats are useable and capable of turning out excellent product, none of them are perfect. I'm sorry Steve, the perfect playback system has not been invented yet.

I never said it was perfect. I only said it plays back exaclty the same thing everytime. Including what you heard as you tracked through the system (not what was in the room)

Modern day analog and digital systems do not reproduce "real world" dynamics or frequency response and all exhibit some undesirable side effects.

There are definite limits to what digital can do, but what ever the limits are, dynamics and frequency response are not limited by current digital technology. Other stronger suspects are sample clock jitter, engineers tracking too hot and over driving the mix bus, Poorly designed EQ's and dynamics, reconstruction filters, anti aliasing filters, dither algorithms, etc

Add to the mix that different individuals hearing varies and it is impossible to ascertain what the absolute highest frequency humans are capable of hearing is. All of this doesn't even take into account that there is a possibility that higher super ultra high frequencies may generate a "fold back" effect and excite lower audible frequencies. All the facts are still not in, the data is incomplete.

Your last sentance is dead on. So far there has been no credible study to show the importance of supersonic frequencies. That does not mean they are not important it just means that so far no one has been able to provide tangeble proof.

Part of the art is in making these systems sound like the “real” thing or at the very least, believable. I think that compression is a valuable tool in this process. I prefer to compress a bit at the time of tracking and then perhaps a bit more at mix.. I almost never used compression across the 2-bus when I worked with an analog console but I have to admit I am doing this more now that I am working in DAW. I have no explanation as to why, it just seems to deliver a better product that way.

Many engineers ran hot levels to tape in order to get the wonderful harmonically rich sound of slightly overdriven tape. Which served as an additional stage of comression. You can't do that to digital, its better in digital to stage a couple of compressors in series and go easy. It also much more of "the sound" of today's music than it was a few years ago. So you might be adding more just to keep up.

I hate over compressed music. Even though I am the “compressor pig” of the West Coast, I still like music with dynamics.. Much like my not being able to drink coffee without a spoon of sugar but not being able to drink it with 2 spoons of sugar, compression is a wonderful thing in small amounts.. Kurt

I am with you all the way on this Kurt. But I will do what the customer asks if needed.

OK- Here's the intermediate perspective- not newbie, not rich audiophile. Either I want to control dynamic range, or I want to use compression as an effect. I'm tracking into a standalone, and I'm not going to use its highly questionable compression software. Track-by-track compression will be done by the nice mixing engineer, with just the kind of high end outboard gear that's been listed here.
Problem is, I need some compression on some signals going in, generally not more than 3:1. If I just want to control dynamics, I use RNC, because it's the best clean compressor I can afford. If I want compression as effect, mostly on vocals, I use a Joemeek twinQ, because it's the best colored preamp I can afford. Plugins are not relevent to me, no PC in the signal chain.-Richie

If someone is nervous about overcompressing on the way in, mult the pre's output and print a compressed and an uncompressed signal.

What I gained from this thread is the idea of doing light compressing on the way in and adding a few dB more in the mix. That is a great idea to get a smooth sound.

BTW, and anyone can shoot me for saying this, but I have an 1176LN and a UAD1, and I have done comparisons of the two on vocal tracks, and honestly I can't hear a quantifiable difference.

ok,
so the looks of it ..

there are lots of nice comps out there for tracking vox.

In the nyc r&b and rap scene I have come across the SSL 384 a whole lot. ( i dont engineer that stuff, but a few friends do ). At giant in nyc the 384 was the sound for a lot of major rap artists.

There were a lot of avalon 737's around in the late 90's too.

Another R&B standard was the sony 800G mic and a manley pre. From there it went to the ssl comp.

My fav is a (clearmountain style) modified la3a. on the way in. I ride it at least 2 dB down and let it dig 16 or 20 down in a loud ass rock chorus.

Another note:
I almost always have a different mic pre gain for verses and choruses. Especially with female singers. They always clip the pre on the chorus. the rest of the chains gain structure stays the same, the mic pre gain comes back 6dB most times.

The SSL thing is that it's really clean makeup gain. On rap you need to leave a lot of headroom at the mic pre. you can get a lot of gain back from the ssl. It adds no noise. and for rap it can grab a lot of peaks without sounding over compped.

I noticed some folks in this thread don't like the sound of their PT or DAW when it is running a lot of plugs. Personally I think that is probably the fault of the plugs, but there is a very simple solution: Render the tracks with the plugs before mixing.