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Author Topic: Digital tracking with low levels = better...is this new???  (Read 147577 times)

Thomas Lester

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Re: Digital tracking with low levels = better...is this new???
« Reply #45 on: January 12, 2007, 09:03:00 pm »

cerberus wrote on Fri, 12 January 2007 18:18

protools may be a popular choice, but perhaps it isn't state of the art for sonics.  for example: with other daws,  it can be nearly impossible to clip anything internally.

jeff dinces


And unfortunately, not having to pay attention to what you are doing has created a slew of so called engineers that have no clue about proper gain staging or how to load a mix bus.  Quite frankly, they've dropped the "engineering" out of Audio Engineering.

I'm not saying a floating point system is better or worse, I"m just saying that a great deal of incompetence has surfaced due to the modern DAW.

-Tom

odysseys

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Re: Digital tracking with low levels = better...is this new???
« Reply #46 on: January 13, 2007, 01:45:33 am »

compasspnt wrote on Fri, 12 January 2007 23:40

Haris, read the thread

http://recforums.prosoundweb.com/index.php/f/29/6490/

in its entirety.  Pay especial attention to the last 9-10 pages.

Then if you still have such questions, come on back and ask them.



The link didn't work!
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Good take is all it takes.

organica

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Re: Digital tracking with low levels = better...is this new???
« Reply #47 on: January 13, 2007, 02:11:48 am »

if you're talking about what I think you're talking about......
go to the sticky at the top of that list or
try this http://recforums.prosoundweb.com/index.php/t/4918/13917/
it's proven 2b life changing for me
in a good way that is
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cerberus

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Re: Digital tracking with low levels = better...is this new???
« Reply #48 on: January 13, 2007, 03:12:23 am »

Thomas Lester wrote on Fri, 12 January 2007 21:03

cerberus wrote on Fri, 12 January 2007 18:18

protools may be a popular choice, but perhaps it isn't state of the art for sonics.  for example: with other daws,  it can be nearly impossible to clip anything internally.


And unfortunately, not having to pay attention to what you are doing has created a slew of so called engineers that have no clue about proper gain staging or how to load a mix bus.  Quite frankly, they've dropped the "engineering" out of Audio Engineering.

I'm not saying a floating point system is better or worse, I"m just saying that a great deal of incompetence has surfaced due to the modern DAW.
i agree that lack of education is a reason for bad engineering.  

however i believe that a floating point system more closely resembles nature, and analog systems.  i would need to be thouroughly re-schooled to  be convinced that i am wrong here:

i believe that we are better off making decisons by listening to the musical results of all our actions, not watching for whether a red light is on or off!   that is bogus toil; and as devo has said "toil is stupid"!    

the computer ought to be working for us, it should be helping to remove technical hassles, not presenting it's own limits and problems as new stumbling blocks for engineers.  

but that is what happened when digital came into use, because at first they were all based on fixed point maths. (and some still are, thus this discussion).

i think that lack of imagination is another reason for bad engineering... "by the book" "paint by numbers" .. sure i'll tell you what frequency is "presence".. but who cares if nobody is buying it?   yeah, we have problems.

fact: analog gpes to eleven.  if we want to emulate analog-like responses, then our system has to go to eleven, all the way... no bottlenecks, none of that bullsh*t is necessary.

why keep your signal below -6?    you protools users are acting like chicken littles!   what is so scary about the uppermost bit of a 24 bit recording?  why should the ones and zeroes contained in the "most significant" bit be less valuable to us than the data from any of the other bits?   bits=information... music, that is.

a.f.a.i.a.c...it's all part of the music, once the signal passed the a/d converter, it would be a crime to throw any of it away. as if one ought to always drop a bit,  just like tossing out the trash? so the new advice is to start mixing with only 23/24 of the signal that was captured? that advice seems fundamentally backward to me.

float can go past  eleven. fixed cannot go past ten.  no sales person is likley to change that.

jeff dinces

tom eaton

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Re: Digital tracking with low levels = better...is this new???
« Reply #49 on: January 13, 2007, 08:34:37 am »

Jeff-

You can't be serious.  Analog has all kinds of bottlenecks.  I can clip ANY analog device.  It can be ugly.  Some analog devices go to 11... some go to 7 and then sound like crap.  

You seem to be suggesting that we throw out the math and theory and ignore the technical aspects of recording while saying that "lack of education is the reason for bad engineering."

I agree that is would be great if the machines got out of the way of recording... but until that happens we need to be aware of how to interface them with the world in the way that best suits the work at hand.  

Using reasonable levels is just one way we can make our tracks "travel" well in the real world.

-tom

Thomas Lester

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Re: Digital tracking with low levels = better...is this new???
« Reply #50 on: January 13, 2007, 12:54:49 pm »

cerberus wrote on Sat, 13 January 2007 03:12

i believe that we are better off making decisons by listening to the musical results of all our actions, not watching for whether a red light is on or off!   that is bogus toil; and as devo has said "toil is stupid"!


I couldn't disagree more.  When I started recording, the most "digital" thing in the studio was an Atari ST, an SP12 sampler, and a 16 bit DAT machine.  The idea of listening exclusively to the musical results to determine our actions is a really bad idea.  In the analog world, just like in the digital world, if I listed to a guitar track by it's self.  I may think to myself...   "It sounds good, so it must be good".  But then, I add 5 more guitars, a bass, two lead vocals, 8 BGV's, a keyboard, Hammond, and acoustic.  

But now...  I've added all those tracks... all with their own small... maybe un-hearable noise floor.  They all multiply on each other to make a horribly loud noise floor.  As an amateur engineer..  I'm baffled to why this happened.  Or worse... I still don't notice it, but a week later, the client calls me up to say that they are at the mastering studio and that after the ME gets it "loud" the noise floor is so obnoxious that they are going to have to start over from scratch.

However...  as a well trained and experienced engineer, I know a noise floor exists whether I hear it or not.  I also no that when two signals combine, they increase in volume.   So, I know to maximize my signal to noise ration as much as possible when recording.

Same goes for digital.  I need to know how the math is working.  I need to know what it's limitations are.  I need to know what the buss (sorry William...  I come from the "buss" camp) does when pushed vs. not pushed.  32bit float his limitations.  48 bit fixed (PT) has limitations as well.  I need to know what they are.  I've also got to get this signal back out somewhere.  I have a very definite limitation on my output D/A.  I need to know what that is, too.

Again...  one clipped signal may not sound bad...  you may not notice it.  But 32 clipped signals is going to start to sound pretty harsh.

I'm also a professional..  that means I have paying clients.   I need to know how this is going to work, because I have to work fast.  They are paying by the hour.  I don't need to realize halfway through the day that my mix buss is getting smeared because I'm stressing the math.  Or that I'm way to hot for my output D/A and I need to make my clients wait.

If you are working on your own stuff at home...  have at it.  Experiment all you want.

Quote:

i think that lack of imagination is another reason for bad engineering... "by the book" "paint by numbers" .. sure i'll tell you what frequency is "presence".. but who cares if nobody is buying it?   yeah, we have problems.


Don't confuse knowing what you are doing with having a lack of imagination.  Knowing what you are doing and what the gear is doing sets us free to be full of fantastic imagination.  Nothing ruins creativity like like stumbling upon and unknown and not knowing how to fix it because you don't know how the system, the math, the electronics, etc. works.

Quote:

fact: analog gpes to eleven.  if we want to emulate analog-like responses, then our system has to go to eleven, all the way... no bottlenecks, none of that bullsh*t is necessary.


Digital can go to "11" as well.  You just have to know where 11 is.  Don't for a second think that Analog doesn't have a ceiling.  It does.  And the thought that it doesn't also appears to be a trend amungst amateur amongst.  Hit an API too hard....  you'll see what I mean.

Quote:

why keep your signal below -6?    you protools users are acting like chicken littles!   what is so scary about the uppermost bit of a 24 bit recording?  why should the ones and zeroes contained in the "most significant" bit be less valuable to us than the data from any of the other bits?   bits=information... music, that is.

There is no "more significant bit" than another.  

Quote:

a.f.a.i.a.c...it's all part of the music, once the signal passed the a/d converter, it would be a crime to throw any of it away.


That's just crazy talk.  Unless you have a track with 144 dB of dynamic range or better (which doesn't exist in the musical world), then you are dropping bits on one side or the other.  Heck...  from the threshold of hearing (for an infant that hasn't had hearing loss) to the threshold of pain, is only 120 or so dB.  So, even dropping the top 6 dB is going to give you 18 dB more dynamic range that human ears can even handle.  

And that's just from the A/D and D/A.  The PT internal mix buss has 288 dB (48 bit) of dynamic range.  So internallly...  dropping 6 dB still gives you 162 dB or more dynamic range than you'll ever need.  On a 32 bit float, you have 192 dB total (or 66 dB more than you'll ever need).  

So...   drop 6 db and be "safer" on my clipping (and if you are tracking... there aren't any floating point A/D converters) and not give up any dynamic range.  Or push the envelope to get one more dB and probably end up with cumulative distortion that causes "harshness" and confusion in your mix.  Your choice.

Floating point has an expense associated with it, too.  Each time it has to shift it's bit range up or down, there's potential for loss.  It's not 100% efficient.  Just like multi-processor machines.  Each time a processor has to shuffle off the work load, it loses efficiency due to the math involved.  This is fact.  I'm not saying that the costs outweigh the advantage.  I'm just saying it's not a flawless system.... but that's not what this conversation is about, so I'll get back on track...

BTW... I, just like you, was taught from the beginning of my career to saturate all the bits.  But that was also when things were 8, 12, or 16 bits.  The max dynamic range of digital back then was 96 dB and the converters sucked and were noisy... so you pushed it.  I thought this way up until we started discussing this a couple of months ago.  I know the math, so I understood Terry and WW's arguments.  It made since, from a physics stand point, so I tried it.  Sure enough...  it made a HUGE difference in my recordings.

BTW... I often go between PT and Nuendo.  75% of my work is PT, because I prefer it, but I do a lot of work out of a room that only has Nuendo.  I tested this on both platforms.  It sounds better to keep plenty of digital overhead on both systems.  There's two reasons for this...    one, the A/D converters are not floating point.  Two, if I'm not making the floating point math kick in, then the system has to work less and less chances for error.

OK... that wa a long post, but I had to weigh in.

-Tom

PS.  Please understand that the comments about amateur engineers has NOTHING to do with you (cerberus) and I am in no way implying that you are amateur or don't know what you are doing.  I have no idea what your skill level is, so it wasn't remotely aimed at you.  I'm strictly arguing the post.    Very Happy

Buzz

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Re: Digital tracking with low levels = better...is this new???
« Reply #51 on: January 13, 2007, 01:47:20 pm »

I'm just curious I have tried recording at the prescribed -18/-20bd to test this idea , at least in my DAW ( it uses a 64 bit fixed mix buss ) I can't tell a difference between -6 and -20 ??? , all the punch and clearity is still there to MY EARS !! ( and there are'nt the best out there but not bad either ! ) this is with plugins inserted etc,

Anything to a 64 bit fixed mix buss ????

Later
Buzz

PS: I normally am @ -6/-10db recording levels

Thomas Lester

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Re: Digital tracking with low levels = better...is this new???
« Reply #52 on: January 13, 2007, 02:00:35 pm »

Buzz wrote on Sat, 13 January 2007 13:47

I'm just curious I have tried recording at the prescribed -18/-20bd to test this idea , at least in my DAW ( it uses a 64 bit fixed mix buss ) I can't tell a difference between -6 and -20 ??? , all the punch and clearity is still there to MY EARS !! ( and there are'nt the best out there but not bad either ! ) this is with plugins inserted etc,

Anything to a 64 bit fixed mix buss ????

Later
Buzz

PS: I normally am @ -6/-10db recording levels


Safe is safe...  you are probably running safe levels at -6 peaks.  So, -10 peaks will be "just as safe".  Try it with getting as close to 0 dBFS without clipping, then compare it to staying under -6 dBFS.  Then you should hear a difference.  

Also...  don't try one instrument or track.  Track a 16+ track session both ways (as close to 0 dBFS and under -6 dBFS), do a full mix of both.  Then compare... but compare more than just the finished product.  Compare how long it took you to get to a finished mix.  Did you have any issues?  Did you struggle more on one than the other?  How did you ears feel after then mix was complete?  Did you have to take more breaks than usual?  How do the final mixes compare?


PaulyD

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Re: Digital tracking with low levels = better...is this new???
« Reply #53 on: January 13, 2007, 02:39:34 pm »

Not to belabor the point about dynamic range, but...keep in mind the dynamic range of human hearing is only about 120 dBSPL - and that's someone with good hearing. Those of us who spent a good portion of our youth camped in front of drums and guitar amps turned up to 11? Yeah right...

And yes, I know dBFS does not necessarily equal dBSPL. But unless you're monitoring at 120 dBSPL (I hope not!), then you're getting even less dynamic range on playback. Then we start compressing and limiting things...We, as a species, aren't necessarily in love with gobs of dynamic range. So...don't be afraid to record between -12 to -18. As has been said, it will sound better.

Also, it is not difficult to clip a 32-bit floating pointing DAW, either between plug-ins or in summing. I've seen it plenty of times, using both pre-fader metering and spectrum analysis plug-ins. As a matter of fact, I've seen plug-ins with absolutely no level controls whatsover cause clipping, regardless of any of the other settings used on the plug-in. They were just programmed that way. No kidding.

Read Nika's Book.

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compasspnt

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Re: Digital tracking with low levels = better...is this new???
« Reply #54 on: January 13, 2007, 06:33:04 pm »

Another thing to consider is that in most cases, even the best converters today have a S/N ratio of 120 dB.

If digital recording gives you 144, then, even if you are consumed by the S/N ratio, you still have 24 dB to play with.
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organica

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Re: Digital tracking with low levels = better...is this new???
« Reply #55 on: January 18, 2007, 12:44:54 pm »

fwiw .....
I mixed a record last summer for a band which worked for  them I guess
they had tracked it
I remember talking to them about bringing the recording levels down in the future

they just brought me some more stuff to mix where they've done just that
the difference is huge !
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maxdimario

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Re: Digital tracking with low levels = better...is this new???
« Reply #56 on: January 18, 2007, 01:21:27 pm »

Quote:

If it's truly a floating point plugin, it will have a gain range of about 1500dB (at 32bit floating). There's about 512dB of headroom above 0dBfs, and about 1024dB going downwards.

If the plugin is clipping, then it must not coded properly. I did plenty of tests in the ol' trusty SADiE system, and I couldn't clip any of their plugs.

When it comes to fast transients, that's where floating point loses a bit in terms of math, but fixed point wins out a bit.

Bob, can you give some examples of floating point plugs that clip? I've never seen this! I'm curious and worried!

Cheers,
JB




Since no one answered on the other thread I am posting this quote and would like to know what this is about..(regarding transients)

I (and a friend of mine with good ears) have always noticed that drums lose their impact in digital..

In addition I have the feeling that the levels being too high and damaging the overall sound is NOT due to clipping but due to some dynamic process in the algorhythm of the DAW.

it may be that lower signals are indeed processed at a different resolution...or using a different process...

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Tomas Danko

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Re: Digital tracking with low levels = better...is this new???
« Reply #57 on: January 18, 2007, 05:05:56 pm »

Today we were analyzing a track, figuring out howcome it sounded like it had less resolution compared to the rest of the tracks on the album we're working on right now.

The vocals sounded harsh, closed-in and veiled. I could hear distortion artifacts in the kick drum sometimes.

Then I looked at the master fader, positioned at zero. And it showed -2.8 dBFS peak for the entire song.

Every other track I've been working on shows like -5 to -8.

We lowered the fader 6 dBFS and upped the DAC-1 converter the same amount.

Suddenly there was nothing wrong with the vocals. And no distortion in the kick drum.

Somewhere down the line the master fader got zeroed, from its initial -6 dB position. I recall doing some quick and dirty "mastering" to check the track out suddenly had it sound very edgy and harsh a couple of days ago.

No wonder.

It really is amazing, what a difference it does. And the meter never went over zero, so as far as the computer nothing was broken...
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tom eaton

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Re: Digital tracking with low levels = better...is this new???
« Reply #58 on: January 18, 2007, 06:53:20 pm »

maxdimario wrote on Thu, 18 January 2007 13:21



Since no one answered on the other thread I am posting this quote and would like to know what this is about..(regarding transients)

I (and a friend of mine with good ears) have always noticed that drums lose their impact in digital..

In addition I have the feeling that the levels being too high and damaging the overall sound is NOT due to clipping but due to some dynamic process in the algorhythm of the DAW.

it may be that lower signals are indeed processed at a different resolution...or using a different process...




Let me suggest that drums maintain more of the initial transient in digital.  How you respond to this is anyone's guess.  I think that tape (used in the manner most would use it-- not Massenburg style) typically brings the transient down relative to the "body" of the sound... making the sustain portion of the sound louder relative to the attack.  In my experience putting some iron (transformers) in front of your a/d converter can help get back a little bit of that sound.. providing some transient distortion/glue/whatever you'd like to call it.

You should consider that clipping IS a dynamic process.  The Frindle posts that are in the archives have TONS of info on how signals are handled internally.  No particular black magic.

-tom


eightyeightkeys

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Re: Digital tracking with low levels = better...is this new???
« Reply #59 on: January 18, 2007, 08:52:26 pm »

We need Dan Lavry in here to straighten this transient thing out.

But, personally, I think it is all in how you initially capture the sounds-the transients, etc...should be perfectly preserved, if done the way you need it -right from the beginning.

A good example of this are drum sound libraries/VSTi's
Stylus RMX, for example. There are many really excellent sounds in there with snappy transients and some with exceedingly fat bottom end, then there are others that are wishy-washy and everything in between. Each sound for a purpose.
If any of these hundreds and hundreds of sounds were recorded to tape first to preserve the transients, I'd be floored.
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