Passive Volume Control - Pros and Cons (Was a PT vs N4 thread) [Archive]

RSE

07-15-2009, 04:51 AM

After the huge investment in PTHD that virtually demands a preference justification from a psychological standpoint alone, I'd be curious to know what you'd use Nuendo for in a heartbeat (since being familiar with it presumes you know the areas in which it is superior)?
Not trying to start an argument in any way, just curious.
Many of us use Nuendo specifically for its features AND the fact we don't like the idea of being locked into a proprietary world... that's our axe to grind. :wink:

It stands to reason that these tools allow so many workflow possibilities that many people with just as much experience as you who are equally familiar with both would choose Nuendo as their personal preference, so your personal reasons for the "PTHD-in-a-heartbeat" would be welcome to possibly shed light on things we might not be considering.

ATM I prefer Cubendo for 'midi and utility' work + 'project studio' stuff - features like ASIO, studio devices, macros and generic remote to name a few make it very easy to interface with 'non-propriety' hardware and software and manipulating data - as long as I have time on my hands and I'm dealing with small/remote jobs, recording one/few tracks at a time (e.g. recording at some client's project studio).

The thing is that Cubendo is loosing by the minute, because Steiny's designers wouldn't recognize a decent studio system even if it bit them in the ass.

I'm so ancient that my first Digi DAW was 'Sound Tools', my first Steiny DAW was 'Pro 24' and my first Emagic DAW was 'C-Lab' (I'm talking 80's here).

With every PT release they integrated more good features they found on other software (including Cubendo), while Steiny's designers keep on adding useless features and not supporting (if not ditching altogether) the features they worked so hard to include in previous versions (take for example the 'control room' that no pro would touch and the generic remote that is buggy as hell) -

A gotta-have feature like tab-to-transient is not on their priorities list, even simple features like (very limited) plug-in/sends drag-and-drop or (very limited) side-chaining appeared only recently in Cubendo 4 (!), and even then they were poorly and partially implemented, if one compares to the 'industry standard'.

I chose Cubendo as my home mixing DAW mainly for the reasons you've mentioned, + PT midi was a nightmare THEN - check out PT's midi now...

Just by comparing the routing and grouping methods in both systems it is obvious that Steiny ain't got a clue as to the needs of a recording engineer.

Don't get me started on the the automation...

Sorry, I wish I could say nicer things about Cubendo but I'm running out of compliments with every new release.

Sam

07-15-2009, 06:17 AM

I didn't really want to get into it...but I do this for a living and use PTHD and Nuendo equally everyday....and the control room is one of the things i miss MOST when in PT land!!....creating a quick and dirty headphone feed off an ITB mix with complex routings is a pain for me in PT....let alone being able to quickly mirror my mix levels on aux sends like I can with nuendo....

So your comment about the control room that no 'pro' would touch I really cant understand.(we need to ban the use of the word 'pro' on this forum - i HATE the way that word is used on the web!! lol!)

There are a few big things Nuendo lacks compared to PTHD, and of course the structural difference of the mixer not residing on DSP with near zero latency at all times....there is also many things PT misses from nuendo...simple stuff like not being able to even select a bunch of regions on a track and apply a 10ms fade in and a 1sec fade out to all in one go....AND....i can pull a bigger mix on my lowly q6600 with N4 and a UAD2 quad than I can on the PTHD 3 Accel system on a mac pro....

I use both to their strengths....but I do get very frustrated with the achaic mannerisms of PT many many times a day....I can edit anything bar drums much faster in nuendo, and mixing/routing/creating aux fx and sending to them is soooooo much faster for me in nuendo....accessing a plugin you are sending to from the source send instead of scrolling through 100 tracks to find it....not having to label ALL my busses so i can keep track of what shit is going where....(mostly) intelligent soloing....i could go on...

I actually PREFER the automation in Nuendo to that in PTHD....and in almost every case I find that the Nuendo routing is faster to get set up.....(and last time I checked I thought I was a recording engineer - my needs are met in that department) though I totally agree on the absolute lack of grouping in Nu....

For me at least, there are very many shades of grey right now....but with the way the i7 stuff looks to be heading, and if steini can get groups, session data import and multitrack warping happening properly in Nuendo then I will have very few reasons to keep waiting for PT all my day long....it is stable but slooooow.

thats my 2c from my day to day anyways....

Sam

Daryl

07-15-2009, 06:33 AM

With every PT release they integrated more good features they found on other software (including Cubendo), while Steiny's designers keep on adding useless features and not supporting (if not ditching altogether) the features they worked so hard to include in previous versions (take for example the 'control room' that no pro would touch and the generic remote that is buggy as hell) -

Well I use Control Room, and I'm a Pro, so you're obvioiusly wrong. :rotfl:

A gotta-have feature like tab-to-transient is not on their priorities list, even simple features like (very limited) plug-in/sends drag-and-drop or (very limited) side-chaining appeared only recently in Cubendo 4 (!), and even then they were poorly and partially implemented, if one compares to the 'industry standard'.

Nah, tab to transient is spectacularly useless for me. Never would use it. Horses for courses I think.

I chose Cubendo as my home mixing DAW mainly for the reasons you've mentioned, + PT midi was a nightmare THEN - check out PT's midi now...

PT is still pretty much cr*p, compared with Cubendo. No Pro would use PT for MIDI. :wink:

Just by comparing the routing and grouping methods in both systems it is obvious that Steiny ain't got a clue as to the needs of a recording engineer.

Now I would certainly agree about Grouping.

D

RSE

07-15-2009, 05:14 PM

No pro would touch the CR because:

1. A decent passive balanced volume control is 120$US, and for an extra 120$US one can get all the switches relays and trim pots one needs for the auxiliary monitor systems - all fed from the same pair of converters.

OTOH an Apogee 16x system costs ~6K$US, in a busy studio environment we always need more ins and outs (summing, triggers, recording, sampling etc.) and using half the outputs for monitoring only (biggies, midfield, nearfileds and small speakers for ex.) is not a popular concept.

2. Attenuating one's monitors with digital faders is yet another unpopular concept in a critical listening environment because once the fader is moved away from the 0dB position it's in quantization errors' territory.

---

quote: creating a quick and dirty headphone feed off an ITB mix with complex routings is a pain for me in PT

Option 1: patch a send across the master and route it to an output - there, U're done. U can copy that send to any other master or track - try that in Cubendo.

Option 2: copy track automation (or just position) to any send from any number of selected tracks, from volume/pan/mute/LFE properties, some or all of them. Try that in Cubendo as well.

---

quote: let alone being able to quickly mirror my mix levels on aux sends like I can with nuendo....

Just insert any number of aux tracks and feed the mix to them, for any number of level meters U wish.

---

quote: i can pull a bigger mix on my lowly q6600 with N4 and a UAD2 quad than I can on the PTHD 3 Accel system on a mac pro....

Excuse me, this is ridiculous: If you want to compare systems please use your UAD Quad in PT as well. A Quad card with no extra plug-ins is 1500$US and has the power of 10 UAD-1 cards - I should know, I just bought one.

---

quote: I can edit anything bar drums much faster in nuendo

When performing simple edits speed (bar drums or multiple phase related tracks...) is more or less the same in both systems for me.

OTOH when editing from 20 different edited versions of the same 80 tracks... Oh right, there are no playlists in Cubendo.

---

quote: mixing/routing/creating aux fx and sending to them is soooooo much faster for me in nuendo....

Hmm... In PT for ex. routing a track to 6 outputs is a simple matter of multiple selections in the routing menu. In Cubendo I have to use individual sends (!) for that + set the gain, pan etc.

In Cubendo I can't route a buss o/p to its i/p - for ex. for delay feedback with each repetition going through the original chain.

In PT I can assign an echo send to 40 tracks in one go and copy the faders' automation/balance to that send. How fast can you do that in Cubendo?

---

quote: not having to label ALL my busses so i can keep track of what shit is going where.

You name your return tracks, that's the same thing.

----

quote: I actually PREFER the automation in Nuendo to that in PTHD

I can answer to facts, not preferences.

---

quote: Well I use Control Room, and I'm a Pro, so you're obvioiusly wrong

See above.

---

quote: Nah, tab to transient is spectacularly useless for me. Never would use it. Horses for courses I think.

Funny, this is the most worn out key of the keyboards of the studios I work in.

---

May I humbly suggest that the simplest explanation is probably true:

a. PT is the industry standard for a reason other than 'habit'. Studio owners and engineers are known to think for themselves, and they had plenty of time to do so.

b. Young ppl go for PT for a reason other then 'what they're taught at school'.

Reading the PT manual is a good advice for a Cubendo owner who wants to decide for himself.

TAFKAT

07-15-2009, 05:36 PM

:pop_corn:

kdm

07-15-2009, 06:10 PM

2. Attenuating one's monitors with digital faders is yet another unpopular concept in a critical listening environment because once the fader is moved away from the 0dB position it's in quantization errors' territory.

Technically true, but there is also some audio-elitist voodoo that has propagated this through the industry. If you are a mastering engineer, then sure, but spend the $5k on a monitoring controller.

A volume control that actually doesn't impact your audio more than quantization errors is going to run thousands, not $120, even passive. I seriously doubt any engineer that thinks they can hear quantization error in 4 bits, much less 1 or 2 is going to rely on a $120 passive - more like $3-5k+.

Quantization errors are partly the fear of engineers from ProTools' Mix days of truncating the crap out of everything that left a dsp chip for the tdm buss when you moved a fader.

I used to be more anal about digital paths than I am now simply because I figured out that the difference is so minor in a quality digital system that it was costing me more time than it was making for my studio. So, I have no problem dropping the control room volume when I just want a quieter monitoring mix.

That said, it usually sits at 0db since my system is calibrated for 83dB with 0 on the control room send.

There are plenty of bigger issues in digital than quantization error from the control room fader esp. vs. a $120 control. I had one that everyone recommended, it sucked. I wouldn't spend less than $3-4k now, though I haven't heard BlueSky's $700 controller - might be decent.

Sam

07-15-2009, 07:17 PM

heh....from the way you responded, and missed a lot of what I was actually getting at....I just cannot get into this one again, because I can already feel the venom and disdain in your tone - and the 'busy studio' gives me authority vibe just doesnt carry weight, as you are using like it puts you in a position we are not in. Mine is a busy studio - even if it is just me and a producer with clients in there - it still works 5-6 days a week. And shock horror - I use control room and have multiple monitor sets, and the audio loss is so neglegable myself, nor any client has ever freaked out when I have pulled the fader down a few DB or hit the dim switch to talk to them over a printing mix.....but maybe I am not 'pro' enough to know destroyed audio quality when I hear it. But I do know how a cheap stereo pot can screw your stereo image, fuck with frequency response etc - so for now I still prefer croom until I wanna drop 6k on a good 5.1 monitor controller with bass management etc ect.....the argument about using i/o is kina moot - if the studio is so busy and need i/o - buy more.

How is it ridiculous to compare a PT HD3 Accel system with N4 and a quad - I am simply pointing out that 1 dsp assisted system costing much less aloows me more grunt than another costing much more. Uad works like poo with PT anyways due to the RTAS fudging, not withstanding that RTAS is a steaming pile to begin with. Lets not even talk about VSTi's in RTAS land, let alone setting up multi timbral VSTis in PT!!!

I dont have the energy to go through all your responses and point out how they dont solve my few grievances - though some of what you say is quite correct with reference to setting up multiple sends simultaniously and some of the routing limitations - but in 95% of cases I still prefer the cubendo way, but yes it could absolutely be improved especially with a 'apply to all selected' modifier. But for example - naming returns is not the same as naming busses - in pt i would need to open the i/o setup, rename the buss/s and also name the aux input, set the aux input bus, and then send to the bus....OR in cubendo....create FX track, choose plugin, boom - you have a named bus to send to, and a channel named by the plugin that is on it.

I am really glad you find PT fits your needs just fine - what I dont get is why you feel the need to belittle others professionalism, without knowing squat about them or their work, just because they find cubendo features fit their needs just fine.

Bottom line is, I know what I am comparing side by side every day - and I still need them both so there is no golden bullet for me....anything beyond that is pretty irrelevent.

As you were ;)

RSE

07-15-2009, 09:11 PM

I'm sorry that you don't like my answers but this means that you don't like the facts.

My point was 'leaving price and company policy aside - PT is a better system then Cubendo' - evolving faster and provides better solutions for a busy studio environment. Read my original post:

ATM I prefer Cubendo for 'midi and utility' work + 'project studio' stuff - features like ASIO, studio devices, macros and generic remote to name a few make it very easy to interface with 'non-propriety' hardware and software and manipulating data - as long as I have time on my hands and I'm dealing with small/remote jobs, recording one/few tracks at a time (e.g. recording at some client's project studio).

'A busy studio environment' means working under time pressure with a large number of people - e.g. producer + programmer + 4 band members + girlfriends + a 5 piece string section all at the same time - it DOES NOT mean 'I'm a pro and you're not':

Lo and behold, I'm a pro and I do small (me + 1 client) sessions as well.

---

As for the CR debate: even if you prefer using digital faders for attenuating you're still using up 8 expensive converters for monitoring only - and this is a good enough reason not to use the CR feature -

But FYI I did some a/b/x blind tests before I made up my mind, and I also tested with non-audio people - I even recorded through the bloody thing and compensated the level for comparison: the passive attenuator sounds better, period.

Take a simple test in Cubendo: with any mix, drop your faders by 15 dB and raise the master by 15 dB - and see if you can tell the difference.

FYI, this (http://store.nhthifi.com/NHT-PVC-Pro?sc=12&category=1225) is what I use at home, and here is the street price (http://www.listenup.com/NHT+PVC+Pro-p-PVC_Pro-p-50052.html). FWIW it sounds better then the Neve 8816 (analog) CR level control.

---

While on the subject of listening tests, there's no 'venom and disdain in my tone' and I have no hidden agenda: after all I do use N4+C4 at home (and in some studios) as my main DAW, don't I?

But it is my opinion that as audio professionals we should face the facts, not the politics.

TerryG

07-15-2009, 10:06 PM

As long as you can reconcile your "fact" with your "opinion", and allow others to have theirs as well... :wink:

Habits and occasional explorations determine workflow preferences... if yours were obviously best, everyone would agree, and we wouldn't be here, would we?

Sam

07-15-2009, 10:11 PM

Hey Vin, pass me the coolaide flask buddy....I need to see the 'facts' for what they are :)

kdm

07-15-2009, 10:30 PM

Yes, I have the NHT. It's fine for most uses, but I could tell it had L/R balance issues in the pot, as well as some degradation at low levels, to the point that I found I was only happy with it at its' optimal level, defeating the purpose (and that was before I moved to 5.1, rendering it obsolete for my needs).

Potentiometers suffer a similar problem as we find with quantization noise - resistance is used to attenuate volume and as you decrease the level with the pot, you are also decreasing the "resolution" (for lack of a better term) of that resistance. It's audible in less expensive volume controls.

Your example of dropping faders and raising the master is actually more of a quantization noise problem than a single digital level because you are subjecting multiple tracks to individual instances of bit reduction (albeit still in 32-bit float in Nuendo), not just a single audio stream at the final output buss.

Also, the only true solution to avoid quantization noise/errors is to mix analog and run all DAW tracks and faders at unity gain - e.g. never mix within ProTools or Nuendo, and never use fades, crossfades or clip gain. But then you add analog noise issues, and completely defeat the reason for using a DAW.

We all know the gain staging concepts in both the analog and digital worlds - nothing new there, but I finally reached the conclusion that obsessing over quantization noise, and similar issues ($5k power cables, balanced/gold connected wifi access at Starbucks, etc) was hardly a productive quest when comparing the speed and efficiency of switching between my 5.1, stereo and mono feeds, the $5k it would take to move that to hardware monitoring to my satisfaction vs. the efficiency of handling it in Nuendo.

My conclusion after years of these debates and knowing what I hear vs. what people say they hear, is to do what works best for the music/audio, my clients, and my bottom line, and leave the audiophile/avoid-technology-if-it-isn't-perfect debates to people with more time than I have (not saying you are one of these of course). :-)

Btw, on the PT vs. Nuendo thing, everyone has good points, and their own preferences. I don't think there is an absolute "best" - just a best for some uses, and preferred for the rest.

There are things I love about both apps, and things I hate. Currently Nuendo is higher on my hate list, but everytime I use ProTools, I am reminded why I haven't left Nuendo yet.

LEX

07-15-2009, 10:35 PM

Well, I have to say, with RSE, the NHT Pro Stuff is KILLER.

I have a set of A20's with C20 amps and have used the PVC before (as a loner when my Furman control room was acting up), and it was great.

I'm using the Presonus Central Station, and prefer it over the control room in Nuendo.
Personally, I never liked it.

LEX

Sound Drifter

07-15-2009, 11:17 PM

I'm not sure if I read this or someone told me this and reading this topic I thought of it. Maybe both.

Was PT the first company to offer DSP acceleration cards?

I read wiki, but never fully trust the info there. Wiki said Sound Tools was PT's first incarnation and it consisted of a DSP accelerator card.

The reason I ask is back then in the 80's computers really didn't have the horse power to do all this audio work and digital audio was beginning to surface, since PT had these cards to add the horsepower was why so many people moved that way.

For what it's worth and if anyone even cares, I was at a panel discussion a few years back. George Massenburg openly stated "I like Nuendo, I really like Nuendo." when someone asked him what sequencer he liked working with.

TAFKAT

07-15-2009, 11:22 PM

I'm sorry that you don't like my answers but this means that you don't like the facts...

Let me try a little Diagonalese interpretation if I may..

I'm sorry you don't agree with my pin width view , I am omniscient , just ask me....

:pop_corn:

07-15-2009, 11:25 PM

:pop_corn:

ROCKINROG

07-16-2009, 12:32 AM

:emote_beerchug:

RSE

07-16-2009, 01:35 AM

Let me try a little Diagonalese interpretation if I may..

I'm sorry you don't agree with my pin width view , I am omniscient , just ask me....

:pop_corn:

If by 'Diagonalese' you mean 'diagonalised' then the signature in D's message following yours is very appropriate:

Dreams should never be reduced in potency by subjecting them to reality. - Lord Snarebottom

Besides being a very talented musician, he is probably right about SOME Cubendo users.

....The answers I got didn't deal with (most of) the facts I raised, did they? :D

----

Yes, I have the NHT. It's fine for most uses, but I could tell it had L/R balance issues in the pot, as well as some degradation at low levels, to the point that I found I was only happy with it at its' optimal level, defeating the purpose (and that was before I moved to 5.1, rendering it obsolete for my needs).

The maintenance tech that recommended it to me analyzed it with a scope and found it was up to their spec.

When I got my box we repeated the test - same result.

For 5.1, a DACT CT2 (http://www.dact.com/html/attenuators.html)goes for ~400$US.

---

Potentiometers suffer a similar problem as we find with quantization noise - resistance is used to attenuate volume and as you decrease the level with the pot, you are also decreasing the "resolution" (for lack of a better term) of that resistance. It's audible in less expensive volume controls.

I've never heard of that - can you please explain the technical background for that? Which signal property is degraded exactly?

---

Your example of dropping faders and raising the master is actually more of a quantization noise problem than a single digital level because you are subjecting multiple tracks to individual instances of bit reduction (albeit still in 32-bit float in Nuendo), not just a single audio stream at the final output buss.

Try it with a single stereo fader with 40dB instead.

---

Also, the only true solution to avoid quantization noise/errors is to mix analog and run all DAW tracks and faders at unity gain - e.g. never mix within ProTools or Nuendo, and never use fades, crossfades or clip gain. But then you add analog noise issues, and completely defeat the reason for using a DAW.

1. Dealing with quantization noise/errors in a mix is one thing - we compensate until the mix sounds good.

Sound (unpredictiably) changing when the monitoring level is changing is a different thing - the whole idea is to monitor THE EXACT SAME MIX at different monitor levels:

Fletcher-Munson curves are predictable - digital errors are not.

2. Hiss is not the reason for using DAWs, s/r ratio is sufficient in the analog domain. In fact, on some mixes ADDING hiss makes the mix sound better.

...Some all analog mixes still sound very good, IMHO :D.

---

Btw, on the PT vs. Nuendo thing, everyone has good points, and their own preferences. I don't think there is an absolute "best" - just a best for some uses, and preferred for the rest.

Very true. May I remind you that the original question was 'why do younger audio guys PREFER ProTools'.

RSE

RSE

07-16-2009, 01:47 AM

...btw vca style grouping is possible in Cubendo by using the Project Logical Editor + macros.

Automating them... that's a different story.

RSE

LEX

07-16-2009, 01:48 AM

Why do younger guys prefer protools?

We'll if you look at it as a whole, it is cheaper than Cubase and easier to use.

You get an audio interface and the software.

All the basics of PT work. Navigating is easy. Exchange is easy thanks to import session data.

There is a simple and easy upgrade path, regardless of the "cost" in comparison to Cubase.

It just is a standard. Word of mouth about SB, even Logic puts these new guys off.
They want to do something, no bury their head in a manual for 3 weeks.

Granted, Cubase is pretty straight forward in comparision to Logic.
But, when I was at guitar center this past weekend, all the salesman were doing was pushing Pro Tools.

I even over heard another customer there telling this guy why NOT to get Cubase.

Even with all the different problems PT8 has, though less than any SB release, PT8 is still pretty good.

I'll be on PT 7.4.2 until 8.2 probably. Just like from 6.9, I waited until 7.3.1.

LEX

LEX

07-16-2009, 01:49 AM

...btw vca style grouping is possible in Cubendo by using the Project Logical Editor + macros.

Automating them... that's a different story.

RSE

Then whats the point? :rotfl:

Wouldn't it just be easier to have VCA groups.

Of course, they don't understand that concept.

LEX

TAFKAT

07-16-2009, 01:54 AM

If by 'Diagonalese' you mean 'diagonalised' then the signature in D's message following yours is very appropriate:

Nooo, I meant exactly what I stated.., and of course there is no need for me to define it further , as it would be inappropriate for me to dare question your omniscience..

Don't mind me Mate, I feel privileged just to be party , I'll sit back and allow you to keep educating us.. :eusa_whistle:

Carry On...

:pop_corn:

RSE

07-16-2009, 02:18 AM

I'll sit back and allow you to keep educating us.. :eusa_whistle:

Carry On...

:pop_corn:

Touchy, touchy... :D

colony nofi

07-16-2009, 03:42 AM

oh fuck oh fuck where do i start....do i have time for this now...wtf wtf wtf.
Now I've got that out my system... and since I *don't* have time for this since I have 8 clients sitting in the next room waiting to join me...on nuendo...running on a LAPTOP!!! A coolaid variety at that... damn, the world has gone mad.

There are SO many differences between cubendo and alishad HD that I would almost call them different paradigms...

I use both almost every day. (At the very least weekly...even on holidays....)

AND I use DP, Logic, Reason, Max, Live, 2", energyXT, AudioMulch Blah.

They all work. AND THEY all work in a professional environment, if you know them well enuf (and that includes their weaknesses).

Ah, I really don't have the time.

I will. I really will bite. There is some really f*cked up thinking out there regarding usage of tools....

Back to it.

Brendan...

Daryl

07-16-2009, 08:15 AM

Well having read through this thread, (with the exception of some of the post by someone who doesn't understand how to use the Quote function), I think that it is obvious that that are many different workflows here.

People who work the way I do can't use PT. People who works the way that some engineers like to work can't use Nuendo. Nether studio is more professional than the other. The idea of tarring all Pros with the same brush is narrow minded, and fairly ridiculous.

End of story. :sleeping:

D

07-16-2009, 09:02 AM

Besides being a very talented musician

I should state now that this sort of nose polishing doesn't really hold any weight with me. :wink:

ramagochi

07-16-2009, 11:26 AM

Hi,

No pro would touch the CR because:

1. A decent passive balanced volume control is 120$US, and for an extra 120$US one can get all the switches relays and trim pots one needs for the auxiliary monitor systems - all fed from the same pair of converters.

Add a passive volume system to your monitor can degrade the sound so much (Impedance problems), worst if you use a long cables. Will be work with a very small cable from the pot to the amplifiers or crossovers (Less than one meter maybe). In other hand, you can use these passives with a good buffer for use a long runs of cable without impedance problems.

This is a typical question in the hi-fi high end forums, you can get more info in Google using: "impedance problems of the passive preamps"

2. Attenuating one's monitors with digital faders is yet another unpopular concept in a critical listening environment because once the fader is moved away from the 0dB position it's in quantization errors' territory.

:icon_lol: with 24bits your quantization error is on -144dB range (I doub that you can heard this), lower than the resistor noise. You lost 1bit every 6dB down (-48dB for a 16bit level)

Cheers
Suso

kdm

07-16-2009, 01:18 PM

I've never heard of that - can you please explain the technical background for that? Which signal property is degraded exactly?

To be honest, it's too technical to get into here (analog circuit theory), and I don't have time to put everyone to sleep with it. :wink: I left that world years ago for the more creative music world. Just suffice it to say that a clean analog circuit is quite expensive to design and not as accurate as most want to believe, even a passive one (even if it passes your signal test one day - test it again in a year and be sure to run the full rotation of the pot as well, not just your listening level).

Then just the mechanical limitations of a potentiometer can lead to many problems (dust over time, corrosion/oxidation). Even a little can degrade the signal - the idea of putting a variable connection between your high end converters and monitors is basically 2 steps forward, 9 steps back. We do it because we have to, not because it's in keeping with the accuracy requirement of our monitor chains (not to say the digital solution is better, just that analog isn't a magic bullet either).

What I can tell you from real world use is I heard imbalances and inaccuracies in the Left and Right channel balance with the PVC esp. at low volumes - enough that I dropped it fairly quickly. A pot's taper becomes less reliable at lower, and often higher levels (the lower the resistance for the same current, the lower the voltage/amplitude; and converse for higher resistance).

Sound (unpredictiably) changing when the monitoring level is changing is a different thing - the whole idea is to monitor THE EXACT SAME MIX at different monitor levels:

And you really think a variable resistance path is "exactly the same" across the range? Far from it. This is the nature of pots. It's very costly to make one consistent across the range.

Anything analog in your chain removes the option of calling it "exact", and analog circuits vary over time, with heat, and with power fluctuations, even if you have a power conditioner. The best we can ever do with analog is to closely approximate a digital signal, which is ironic since that was the same accusation leveled (accurately) against digital years ago. It's still true, but the point is that you are introducing a variable into the chain with an analog control. Likewise, severe digital level changes do reduce bit resolution.

The point isn't to throw out analog volume controllers, but to realize that you aren't necessarily going to meet the expectation of true accuracy with a $150 controller. It just isn't that simple, and unless you make severe level cuts, quantization noise isn't as much of an issue as low tolerance resistive pathes in your monitor chain, as ramagochi pointed out.

2. Hiss is not the reason for using DAWs, s/r ratio is sufficient in the analog domain.

No one said it was *the* reason, but it's one reason I went digital years ago. Try this - listen to an analog recording of an orchestra with the violas playing ppp compared to a digital recording (if you can even hear it in the analog recording above the noise floor).

In fact, on some mixes ADDING hiss makes the mix sound better.

Eh... that's only because you are masking something you don't like in the digital recording. As a rule though, it only adds noise, which we spent much of our time in the analog realm trying to minimize. Why go backwards?

As far as dither, the reasons there are generally sound, but also somewhat overblown in some cases.

As far as analog vs. digital recordings, other than for pure musical reasons, I have yet to find an analog recording I prefer over a quality digital recording - performance, material and engineering quality being equal. Not to say all digital work is better - some is poor, and some engineers are poor at it - much the same as it was with analog.

(-48dB for a 16bit level)

A little math miscalculation: it's -96dB for 16 bits. :wink:

ramagochi

07-16-2009, 03:05 PM

Hi,

A little math miscalculation: it's -96dB for 16 bits. :wink:

Sorry, probably I couldn't explain well, I only said the difference between 24 and 16 not the value of 16 bits:
24bit 2^24=16777216, 20x log16777216 = 144,49dB.
16bit 2^16=65536, 20x log65536= 96,32dB

144dB-96dB=48dB
1bit=6dB (20x log2= 6)
24bits-16bits=8bits=48dB

In resume, if you rest 8bit to your 24 bit mix, you have a 16 bit mix, -96dB of dinamic range and will be an audible quantization error without ditter, but a -48dB in your monitor output is a lot.

Cheers

ramagochi

07-16-2009, 03:27 PM

Sound (unpredictiably) changing when the monitoring level is changing is a different thing - the whole idea is to monitor THE EXACT SAME MIX at different monitor levels:

Your ear isn't lineal at all sound pressure levels (Isophonic curves)

http://www.decimin.com/Images/equal%20loudness-1.jpg

You can't hear the same mix at different monitor levels, you need a correction (The infamous loudness button on the hifi devices) (Better, variable with the SPL), this is one of the reasons that make that be a good idea using a calibrated monitor system, like the Katz system.

Usually mixing at very high SPL levels we tend to down the bass, exactly the inverse occurs mixing at low SPL levels (And of course the use of the dynamics)

With a complete pasive control (Such as switching potenciometer units) between your DAC and amplifiers, the variable impedance in the different positions, especially the high impedance output of these passive units, make it very critic with the wire and will be alter the sound at different positions, you need a buffer after these pot for do well the job, brands such as DACT sell too very good buffers for his pots.

Cheers
Suso

kdm

07-16-2009, 03:34 PM

Sorry, probably I couldn't explain well, I only said the difference between 24 and 16 not the value of 16 bits:

No problem, I must have misread your post - that makes sense. Thanks for posting the equal loudness graph and response on impedance/buffering considerations. Great info for audio engineers to keep in mind.

LEX

07-16-2009, 03:35 PM

:pop_corn:

kstevege

07-16-2009, 05:10 PM

But for example - naming returns is not the same as naming busses - in pt i would need to open the i/o setup, rename the buss/s and also name the aux input, set the aux input bus, and then send to the bus....OR in cubendo....create FX track, choose plugin, boom - you have a named bus to send to, and a channel named by the plugin that is on it.

A week ago I would not have known what this was all about. But I totally understand now, after pulling my hair out trying to understand old school routing, how Pro Tools bussing works compared to Cubendo. LOL. Some here may recall my thread about this comparison a couple of weeks ago.

TAFKAT

07-16-2009, 06:24 PM

Hey Dedric and Suso,

Thanks for the information guys, seriously good stuff.

Now we just need to wait for our daily dose from our omniscient friend... :eusa_whistle:

:pop_corn:

RSE

07-17-2009, 03:39 AM

Now we just need to wait for our daily dose from our omniscient friend...

...Before I go into the issue of 'passive phobia', let's do the math shall we?

:eusa_think:

An Apogee 16x DA (http://www.mercenary.com/apda16chdaco.html) (16 d/a converters) is 3250$US, so using the Steiny CR with digital faders and four monitor systems will set you back 3250/16 X 2 = 1625 $US.

Using the Coleman Audio M3A MK2 Active Control Room Monitor (http://www.mercenary.com/noname.html) will cost you 850$US + 3250$US /(16/2) = 1256.25 $US.

Btw, the same Coleman Audio Control Room Monitor PASSIVE version (http://www.mercenary.com/coaum3mk.html) costs 950 $US - why would anyone pay MORE for the passive version? The simple answer is because it sounds better.

Using the NHT PVC pro (http://www.listenup.com/NHT+PVC+Pro-p-PVC_Pro-p-50052.html) will cost you 79$US +~120$US + 3250$US /(16/2) = 605 $US...

1625 $US...
1256 $US...
605 $US...

Your friendly neighborhood omniscient friend may tell you that 605 $US is less then 1625 $US and 1256 $US, but we are all pros here and we all have our own methods and ways of interpreting data - so look at the figures again and choose whatever looks cheaper to YOU :eusa_think: :D

---

Btw,

similar issues ($5k power cables, balanced/gold connected wifi access at Starbucks, etc) was hardly a productive quest when comparing the speed and efficiency of switching between my 5.1, stereo and mono feeds, the $5k it would take to move that to hardware monitoring to my satisfaction vs. the efficiency of handling it in Nuendo.

a decent 5.1 solution (http://www.mercenary.com/splsumoco.html) costs only 885$US.

---

Passive phobia:

This is a well-known phenomenon, similar to the 'unbalanced phobia' common with fresh assistants who watch me choose unbalanced connections over the balanced ones.

Passive volume controllers DO sound better then active ones, because the non-linearities, noise and distortion of an additional amp are avoided.

Impedance should be kept within range of course, and short low capacitance cables should be used - but we use those anyway, don't we?

The output impedance of good converters like the apogee is ultra-low anyway, and it's the impedance matching and distance of the volume control and the power amp that matters most (more then the converter > volume control), so if placed near the power amp for example (as in powered monitors) there should not be any problem.

...and in the few special cases (i.e. long cables) we just add a line driver/buffer (100$US).

I already have an active monitor control in my Neve 8816 mixer anyway, yet I still spent the extra 79$ for the NHT - why? because it sounds better.

Cost saving and improved sound quality - do the math.

---

Digital faders:

Analog faders sound better then digital faders - just hire/ask for a demo unit and take the a/b/x test vs. Steiny's CR - if you can’t tell the difference then no fader is gonna help you anyway.

Steiny's audio engine uses floating point math (http://en.wikipedia.org/wiki/Floating_point) - this means that the numbers are converted to 24 bit integer AFTER the fader - so it's not integer bit quantization errors you're hearing, it the 'translation' errors, since the numbers don't always 'fall into place', much like sample rate conversion:

When converting SR the math is simple - upsample to the lowest common multiple (http://www.statemaster.com/encyclopedia/Least-common-multiple) then divide back to the required SR -

Does it sound the same? no it doesn't. This means that we can hear 'translation' errors regardless of the 'distance' between adjacent SRs, or (in our case) adjacent levels.

Take panning for example - when mixing ITB there are only three useful positions: hard left, hard right and center. Everything in between sounds blurry and smeared in the mix when compared to the original signal.

With analog panners OTOH it's much easier to find good panning positions, and the left <> right transition is smooth -

But then again, we are all pros here and we all have our own methods and ways of interpreting data, so maybe LCR mixing may fit your needs better :D

---

RSE - 'your friendly neighborhood omniscient friend'

07-17-2009, 03:52 AM

:pop_corn:

kdm

07-17-2009, 04:18 AM

TAFKAT

07-17-2009, 05:11 AM

Man, theres nothing worse than watching a drowning man... http://fc00.deviantart.com/fs37/f/2008/240/6/d/Clutching_at_straws_by_dirtypaintbrush.gif

Hey D, Dedric,

I think we'll be needing something a bit heavier before this race is run..

Here ya go.. http://fc06.deviantart.com/fs17/f/2007/149/4/0/Bong_emoticon_by_kompound_kid.gif

RSE

07-17-2009, 05:25 AM

Take panning for example - when mixing ITB there are only three useful positions: hard left, hard right and center. Everything in between sounds blurry and smeared in the mix when compared to the original signal.

With analog panners OTOH it's much easier to find good panning positions, and the left <> right transition is smooth -

So, in other words (reading between the lines), either you are mixing on an analog console, or you aren't heading your own advice.

...Of course I mix on analog consoles, I wrote it in this thread:

I've been a pro recording engineer for 25+ years, I work in studios with high-end consoles + DAWs, I mix at home as well (N4+C4+PTLE+Neve 8816+analog stuff),

One of the reasons to why I bought the 8816 were the pan controls:

http://www.vintageking.com/core/media/media.nl?id=55623&c=360669&h=8add99f07079bd87e94f

---

Phobia? Really? Do you really think that is a balanced technical argument?

Phobia:

A persistent, irrational fear of a specific object, activity, or situation that leads to a compelling desire to avoid it.

Persistent? yes, and I just proved it to be irrational.

'Phobia' is a balanced technical argument just as 'omniscient' is.

---

I'm not saying, and never did say that analog doesn't have advantages, just that your argument in trying to prove those advantages is flawed. :wink:

Ultimately the highest rate DSD would be a more accurate solution than perhaps even analog as it would give a near perfect representation of any signal, but without the degradation of passing through analog circuits during mixing. Seriously, there is no way analog can maintain a pure signal path - it is by very nature affecting the signal. It works because we learned to make the most of it's strengths and weaknesses. The same applies to digital. It isn't perfect, but it has significant advantages over analog options for most of us, and those advantages far outweigh the disadvantages you seem most concerned about.

I think that I've shown by now that I have no emotional attachment to any piece of gear or software.

Show me anything that sounds/works better and I'll switch in a heartbeat.

That's not the current state of things though.

I only call'em as I see'em.

---

A little something I forgot:

Your ear isn't lineal at all sound pressure levels (Isophonic curves)

http://www.decimin.com/Images/equal%20loudness-1.jpg

You can't hear the same mix at different monitor levels, you need a correction (The infamous loudness button on the hifi devices) (Better, variable with the SPL), this is one of the reasons that make that be a good idea using a calibrated monitor system, like the Katz system.

Usually mixing at very high SPL levels we tend to down the bass, exactly the inverse occurs mixing at low SPL levels (And of course the use of the dynamics)

Please read my posts and kindly quote the whole section, not just the parts you like:

Sound (unpredictiably) changing when the monitoring level is changing is a different thing - the whole idea is to monitor THE EXACT SAME MIX at different monitor levels:

Fletcher-Munson curves are predictable - digital errors are not.

---

With a complete pasive control (Such as switching potenciometer units) between your DAC and amplifiers, the variable impedance in the different positions, especially the high impedance output of these passive units, make it very critic with the wire and will be alter the sound at different positions, you need a buffer after these pot for do well the job, brands such as DACT sell too very good buffers for his pots.

Cheers
Suso

It's not about the impedance, it's about the capacitance - and as I've shown as long as the guidelines are kept the signal is safe:

Otherwise we better call Colman Audio, A Designs, Audioplex, SM Pro Audio and NHT and ask them to remove their products from the market before they do any more damage.

RSE

Daryl

07-17-2009, 05:36 AM

A quick question for RSE:

When you work in Nuendo at home, do you use Windows or OSX?

D

RSE

07-17-2009, 05:48 AM

A quick question for RSE:

When you work in Nuendo at home, do you use Windows or OSX?

D

WinXP SP2 on both machines.

RSE

kdm

07-17-2009, 06:18 AM

Show me anything that sounds/works better and I'll switch in a heartbeat.

RSE

Since the conversation was about the PVC - here are the specs to prove what I said I heard in mine:

"± .5dB interchannel accuracy to -60 dB"

That's up to 1db difference between left and right. I can hear that. Maybe others can't, or maybe your PVC just got a lucky run of resistors. There is a 0dB difference in the audio signal and digital attenuation - yes, there is loss of bit resolution, but which is worse? An imbalanced mix because you inadvertently compensated for it, or a percent or two less accuracy in your bit resolution (which you don't really hear as a frequency or imaging problem, but a lack of depth in lower level signals).

This is one of the problems with resistive passive attenuators - lack of interchannel accuracy, which is exactly the problem I had with the PVC. If you want to stay passive (which has advantages over active), stepped attenuators often give a better channel to channel accuracy.

Research threads on the PVC - there are several pointing out this issue - most users (usually home setups) find it acceptable at normal listening levels, as I also pointed out here, but the downside of resistive attenuation is also laid out by other users and manufacturers around the net (with the same points I've made here). In general, the PVC is good for the money, but it just isn't as accurate as the digital signal you are feeding into it (by way of D/As). As long as you are fine with that, no need to defend your choice by trying to shoot down others' choices that have sound technical merit.

I think that I've shown by now that I have no emotional attachment to any piece of gear or software.

Hmm. Not exactly the impression I got.

Again, if it works for you, great. Just don't come into a thread stating yours is absolutely a better solution. Understand the tradeoffs and you might not get the kind of reaction you did here.

I went the opposite direction with my monitor choice (for now) - I knew the impact of loss of bit resolution and how to account for it, but didn't trust the interchannel inaccuracy of a passive control. Bugged me to hear the shifting of the image, and just leaving it set at a known, calibrated normal position pretty much defeated the purpose.

ramagochi

07-17-2009, 07:02 AM

It's not about the impedance, it's about the capacitance - and as I've shown as long as the guidelines are kept the signal is safe:

Otherwise we better call Colman Audio, A Designs, Audioplex, SM Pro Audio and NHT and ask them to remove their products from the market before they do any more damage.

RSE

Sorry, no. But the capacitance and of course the inductance too help to do the disaster (Parts both of the final impedance).

http://www.kpsec.freeuk.com/imped.htm

For me is a disaster, I tried a lot of these devices and measured his effects by a transfer function on the the power amp output (Complete chain), I had a roll off in high frequency (1,5dB at 15kHz with some wires) and some attenuation in the low bass, and these problems varied depends of the attenuator position, with runs of 5 meters with different brands of wires. I invite you to trying measuring your system with a transfer function analyzer (A complete chain, the reference in your DAC and the measured signal in your amp output {Better load with a fixed resistor than a speaker} and only change the wires).

The pasive attenuators are very old in the hifi side, a brand called Mod Squad (Now McCormack) released a device called passive "preamp" 20 years ago (Or more), nice (I had one), but not for long runs of wire. In the hifi high-end this attenuators have his fans and detractors, will be good for you (an advice) search in google "passive preamps".

Go to the "Passive volume controls"
http://www.dact.com/html/faqs.html

I don't need to call to anything, I know several products in audio that simply don't work like his developers say (as well as the cosmetic industry). My option is easy, I don't buy these products.

But, if you like buy these products, go...

Cheers
Suso Ramallo

P.S. I don't said to you nothing about the isophonic curves, I don't understand you reply, is a reply for me?.

ramagochi

07-17-2009, 08:16 AM

IMHO conclusion:

The studio volume control is a nice addition in Nuendo and very good for a critical listening, if you don't use a very high attenuation levels (No more of 24dB). With 24dB of attenuation you have a 20bit system, with a theoric 120dB of dynamic (Better than most of the real 24bit ADC or DAC's). Isn't possible for the humans hear a quantization error in the -120dB or -116dB range, if you have a studio with a NC15 you need go to the 135dB SPL in mids for hear this quantization error

http://en.wikipedia.org/wiki/Quantization_error
http://en.wikipedia.org/wiki/Audio_bit_depth

http://www.engineeringtoolbox.com/docs/documents/725/nc-noise-criteria-diagram.png

Probably you can hear before the quantization error the aliasing defects of your ADC or SRC

http://src.infinitewave.ca/

Cheers
Suso Ramallo

RSE

07-17-2009, 03:00 PM

Sorry, no. But the capacitance and of course the inductance too help to do the disaster (Parts both of the final impedance).

http://www.kpsec.freeuk.com/imped.htm

Please read my posts: Provided you use good gear with well matched impedance the problem is cable length - the problem with cables is capacitance, not impedance.

---

For me is a disaster, I tried a lot of these devices and measured his effects by a transfer function on the power amp output (Complete chain), I had a roll off in high frequency (1,5dB at 15kHz with some wires) and some attenuation in the low bass, and these problems varied depends of the attenuator position, with runs of 5 meters with different brands of wires. I invite you to trying measuring your system with a transfer function analyzer (A complete chain, the reference in your DAC and the measured signal in your amp output {Better load with a fixed resistor than a speaker} and only change the wires).

...then you were using high capacitance wires. Check capacitance here (http://www.mogamicable.com/Bulk/micr_cables/console_cables/console.html) and compare it to the cables you used.

Anyway the active vs. passive discussion is futile with regard to the Steiny's CR issue:
If you prefer 'active' just throw in a buffer/line driver, it's 100$US, that’s all, problem solved.

Reminder:

Using Steiny's CR is throwing money away because of:

1. Converters cost (no one replied to that).

2. Digital faders (and now I'm being told that even dropping bit resolution with attenuation is 'good enough' when compared to analog circuitry).

---

P.S. I don't said to you nothing about the isophonic curves, I don't understand you reply, is a reply for me?.

Yes, this reply was for you: I wrote the words you've quoted, but you didn't bother to read the original post (http://www.cubendo.com/showpost.php?p=12979&postcount=26). You quoted only half of the section and disregarded the last sentence. Call it 'equal loudness curves', 'Fletcher-Monson curves', 'Robinson-Dadson curves', 'ISO 226 curves', whatever -

They all describe the same phenomenon, and they are all predictable.

When professional engineers listen at different monitor levels they take it into account.

Digital errors however are not predictable, and cannot be taken into account.

It's all written in my post, just read it.

---

Hmmm... No answers yet regarding to the thread's subjest, i.e. Cubendo being "Harder, Better, Faster, Stronger" then ProTools...

If you guys prefer discussing the important issues of cable capacitance / bit length / Kanye West instead - fine by me :D

LEX

07-17-2009, 03:34 PM

Kanye West is a Gay Fish.

:pop_corn:

LEX

blob

07-17-2009, 03:48 PM

ha ha :violinplay3::emote_beerchug::pop_corn:

ramagochi

07-17-2009, 04:43 PM

Hi,

Please read my posts: Provided you use good gear with well matched impedance the problem is cable length - the problem with cables is capacitance, not impedance.

I read your last posts.

My main language isn't the english, and probably I'm writing very short replies in the hope that all the people can understand me. But I thinking that isn't right and need more explain

No, the problem is the impedance mismatch of the system (Out to in), the cables are a part of the problem, and of course the capacitance isn't the only one problem, the inductance is another problem, the resistance is another problem......and more problems....

When you connect a passive potentiometer to an amplifier with a long cable you will have a impedance mismatch, between you attenuator output and the amplifier input, and all are part (Out-wire-in).

If you can't believe me, read the FAQ of DACT

...then you were using high capacitance wires. Check capacitance here (http://www.mogamicable.com/Bulk/micr_cables/console_cables/console.html) and compare it to the cables you used.

I'm using (And tried) Mogami :smash: and Gotham, and Canare, and Van Damme......:icon_lol: exoteric cables as well such as VdH

Anyway the active vs. passive discussion is futile with regard to the Steiny's CR issue:
If you prefer 'active' just throw in a buffer/line driver, it's 100$US, that’s all, problem solved.

Good, very good, this is the right thing to have a nice conversation. In the same way as you can expend your money in anything that you want or you believe.

Using Steiny's CR is throwing money away because of:

1. Converters cost (no one replied to that).

2. Digital faders (and now I'm being told that even dropping bit resolution with attenuation is 'good enough' when compared to analog circuitry).

Converters cost?, why?, don't use you a DAC for the analog output of your DAW?

The CR is cheap by you don't need buy anything more than you own DAC (And this is a very important part for hear our work).

The digital fader are ok for the job, such I had exposed in my previous posts . Every time that you change a level in a track or event you are using a digital fader (But you can send all tracks to an analog mixer for do the job, I made this when I was using my old Protools 3 Nubus and 16 bit converters). But in other hand you can leave the ditter on for a predictable result

Yes, this reply was for you: I wrote the words you've quoted, but you didn't bother to read the original post (http://www.cubendo.com/showpost.php?p=12979&postcount=26). You quoted only half of the section and disregarded the last sentence. Call it 'equal loudness curves', 'Fletcher-Monson curves', 'Robinson-Dadson curves', 'ISO 226 curves', whatever -

I quoted a part of a kdm menssage that wasn't quoted, and my reply was for kdm. My apologies for the confusion, but my menssage was only for remark the imposibility of hear the same mix at different levels.

Cheers
Suso Ramallo

P.S. Fletcher-Munson, or Equal loudness contour, or isophonic curves (I love the iso-phonic name in my country is the most used) as well ISO 226:2003 curves or Robinson-Dadson curves
http://en.wikipedia.org/wiki/Robinson–Dadson_curves
http://en.wikipedia.org/wiki/Equal-loudness_contours
http://en.wikipedia.org/wiki/Fletcher-Munson_curves

kdm

07-17-2009, 04:54 PM

Please read my posts: Provided you use good gear with well matched impeda
When professional engineers listen at different monitor levels they take it into account.

Digital errors however are not predictable, and cannot be taken into account.

Funny that you compare equal loudness to digital errors. You do realize those are two completely different issues that have to be considered separately don't you? I see how you came to the conclusion they could be related in order to support your argument but it's inaccurate to do so. Quantization errors have a completely different impact on audio than the physics of wave transmission and hearing (getting behind the reason for loudness curves, rather than just cutting and pasting them).

And thank you for telling us what pro engineers do! All these years working in this business, after majoring and working for several years in electrical engineering and digital signal processing, and I never knew that.... Gosh, I've learned something every engineering 101 student knows, and natural audio engineers hear instinctively! :rotfl:

Seriously, arguing Fletcher Munson as a counter argument in talking about digital limitations?? Why don't we argue whether it's better to record with a Brauner mic, or eat pasta?

The first time I heard of the Fletcher Munson curve as a point of audio discussion years ago, my first response was - "someone took time to chart the obvious? Okay".

I still didn't see you reply to the L/R balance issue with passive attenuators - guess the random shifts don't bother you, which is fine. And I didn't follow your converter cost argument, or have time to sort through it. I assume you were arguing I/O costs of all digital vs. a console. Not completely accurate since that is highly application dependant (e.g. recording orchestra, bands, working in post, scoring, etc), but I see your point - still apples and oranges though - preference and application requirements are more significant than your less than unbiased cost analysis.

Hmmm... No answers yet regarding to the thread's subjest, i.e. Cubendo being "Harder, Better, Faster, Stronger" then ProTools...

Why reply? You are the one asserting that you have the absolute answers and won't consider the validity of any other perspective. Actually I should remind you that the thread topic is why do younger kids pick PT over Nuendo. We gave our answers early in the thread if you care to read back.

I think you will find that most people here are not only very tech savvy, but also have a very balanced and detached view of the tools we use (otherwise we wouldn't take virtual sledge hammers to them in other threads...lol). We might take one side or the other in a debate thread, but the main reason I do it at least, is to point out an unbalanced or inaccurate approach by a poster. No one ever said that you are completely wrong, just that your approach smelled like defensive, opinionated BS. No offense.

I'm all for a healthy tech debate, but prefer it to take a more balanced view of the pros/cons of all options, not just one.

LEX

07-17-2009, 05:05 PM

:pop_corn:

dcwave

07-17-2009, 05:54 PM

wow 6 pages of stuff. mostly not related to my original post or question. cool I am a controversy starter! :D Carry on.

paulwr

07-17-2009, 06:23 PM

.....damn, I'm back checking in on the progress, and nuthin'...... common, let's hear some more! Actually, I'm deciding whether to get a passive volume control, or just a controller for my master RME control panel. A few years ago I took my master out of an analog mix and went straight out of my Fireface, the the sound improvement was incredible. But is was a crappy mixer and doesn't fit into this debate. Ears I trust implicitly say they just don't hear a problem using the passive approach in their studios, but I have so far shied away from it since my improvement was so great going digital.

Anyway, I'm willing to hear more..............:pop_corn:

TAFKAT

07-17-2009, 06:28 PM

No one ever said that you are completely wrong, just that your approach smelled like defensive, opinionated BS. No offense..

If it smells like , it usually is... :D

There is also a distinct smell of Old Spice in the air, Sam will know what I mean... :eusa_whistle:

:pop_corn:

TAFKAT

07-17-2009, 06:31 PM

wow 6 pages of stuff. mostly not related to my original post or question. cool I am a controversy starter! :D Carry on.

Hey Dave,

I think you should change the title of the thread.., maybe to something like.., " The spirit of Gearslutz invades Cubendo Forum, or something of that ilk " .... :wink:

kdm

07-17-2009, 07:12 PM

Sorry dcwave. Just to recap for those just joining us after seeing the 500 track limit match on your local networks: This is where it started quite reasonably:

One of the comments made me pause a bit - because there is no cracked version of the current Cubendo the younger audio guys are turning to a low cost solution which may turn out to be PT LE.

Curious about what some of you think about that line of reasoning.

...with some insightful and relevant replies.

And this is where it ended rather quickly (no offense):

The reason that the younger guys are moving to PT in droves is because they know that when they get pro they'll eventually buy an expensive system, and far as studio work goes PT is superior, end-of-story.

One opinion to rule them all, one opinion to find them, one opinion to bring them all, and in the darkness bind them.

Apologies to JRRT posthumously for the creative license...

And that's also the reason why so many engineers are far from "pro" in the product they produce. Most just get PT and automatically get promoted to "pro" by association, from the client's perspective. I speak from experience with some of them, some with 30 years in the business (obviously not starting out on PT).

I aim for, and believe I deliver a better product than the PT studios in my area (though they do fine for their market) but battle the mental block of ProTools being the only "pro" solution with potential clients that have little experience outside that realm. That mindset is propagated by opinions just as we've seen here (again, no offense to anyone personally - it's a perception and presentation issue, not a personal issue).

The audio on the CD, DVD, radio, tv, web or film doesn't lend any clues as to the actual source, so isolating one's view of "pro" is far from acting professionally.

If you need PT for technical, preference, or compatibiltiy reasons, by all means invest heavily. If you don't, then invest wisely in what makes you better/faster/more creative at what you do.

ROCKINROG

07-17-2009, 07:20 PM

http://en.wikipedia.org/wiki/Professional :D

TAFKAT

07-17-2009, 07:36 PM

We have discussed this before in the past, IMO the word " Protools " has become synonymous with any DAW to the general populace, many actually think Protools means DAW and the terms are interchangeable, just as Google is used in the same interchangeable manner when many are talking about a search on the web.., i.e, " I googled the answer "..

Now thats a huge thumbs up for Digi marketing on how they managed to implant the term into the urban vocab /mindset , and it is something that we can discuss again , once we get past the focus shift ..

I digress..

:pop_corn:

RSE

07-17-2009, 09:47 PM

Ramagochi and kdm:

I'll explain the cost issue again:

Whatever your needs are, whether you mix ITB, through a summing mixer or a hi-end analog mixer, the number of converters you need buy has to do with the number of different analog streams you play at the same time.

So for example if you only mix stereo ITB, all you need is a single stereo DA converter.

If you mix stereo AND record up to 2 channels at a time you need to add a stereo AD converter.

If you record up to 2 channels at a time, monitor in stereo and feed a stereo headphone mix you need 2 channels of ADC and 4 channels of DAC, and so on and so forth.

Whatever your needs may be, you come up with the number of the DACs that you need - let's name it 'x'.

Whatever that 'x' may be, you need to route your monitor signal to a number of monitor systems, and now you get to choose:

1. Route your monitor signal to a volume control + analog selector that routes the signal to one monitor system at a time

-or-

2. The Steiny way: buy ADDITIONAL converters, each connected to a different monitor system, and switch between them digitally.

These additional converters cost MONEY + you HAVE to use digital volume controls that reduce bit depth AND produce digital errors in a critical listening environment, so while it may appeal to a musician or the recording enthusiast, Steiny's CR is useless to a pro.

---

Funny that you compare equal loudness to digital errors. You do realize those are two completely different issues that have to be considered separately don't you? I see how you came to the conclusion they could be related in order to support your argument but it's inaccurate to do so. Quantization errors have a completely different impact on audio than the physics of wave transmission and hearing (getting behind the reason for loudness curves, rather than just cutting and pasting them).

-SNIP-

Seriously, arguing Fletcher Munson as a counter argument in talking about digital limitations?? Why don't we argue whether it's better to record with a Brauner mic, or eat pasta?

The first time I heard of the Fletcher Munson curve as a point of audio discussion years ago, my first response was - "someone took time to chart the obvious? Okay".

I know what 'equal loudness curves' and 'digital errors' are, thank you.

The comparison is relevant because when using digital faders for controlling monitor levels, both phenomena produce changes in the perceived sound as the volume changes.

-BUT-

While a drop in the perceived high and low frequencies (as studied in the equal loudness curves) is PREDICTABLE, the effect of digital errors on the perceived sound is not.

Engineers can be trained to recognize constant and predicted behavior, so in the case of 'equal loudness curves' changes the engineer can learn how to evaluate the contents of HF and LF based on the perceived level, and make calculated decisions regarding his mix.

But changes due to digital errors produced by digital faders are UNPREDICTABLE, so in this case the engineer cannot learn how to compensate, and so he'll be making the WRONG decisions.

----

I still didn't see you reply to the L/R balance issue with passive attenuators - guess the random shifts don't bother you, which is fine.

I did answer that:

The maintenance tech that recommended it to me analyzed it with a scope and found it was up to their spec.

When I got my box we repeated the test - same result.

On my ADAM S3As and (very) spaced apart auratones the center image is VERY clear, and the NHT keeps it smack down the middle at all levels.

---

We might take one side or the other in a debate thread, but the main reason I do it at least, is to point out an unbalanced or inaccurate approach by a poster. No one ever said that you are completely wrong, just that your approach smelled like defensive, opinionated BS. No offense.

I'm all for a healthy tech debate, but prefer it to take a more balanced view of the pros/cons of all options, not just one.

Funny, I didn't see you react to personal attacks all along this thread which have nothing to do with 'balanced or accurate approach', it's just MY approach that seems to bother you.

So I guess that for you personal attacks in a technical debate are not 'defensive, opinionated BS approach' (No offense).

I don't mind the personal attacks, quite the opposite: they only show any neutral reader the lack of counter arguments -

But your selective preference regarding one approach that by a strange coincidence happens be the opposite view, smells like defensive, opinionated BS.

No offense.

07-18-2009, 12:22 AM

Referring to other professionals as less than such is a general insult, as opposed to a personal one. Is one "morally superior" to the other?

:pop_corn:

paulwr

07-18-2009, 12:36 AM

RSE..........

Look, I'm just a composer who gets by on as much tech info as needed to get the job done...............

I have a machine room and to go passive on the volume control, I'm looking as about a 6' run to each powered amp.

In a way, I'm using passive already, because my Event Precision 8's have an input pad, which I have adjusted to keep my RME control pannel volume control at -20 to -9 db for listening at various volumes, though I generally keep at about an average of that and monitor at moderate levels.

My Cubase 3 sx master out is kept at or near nominal.

I'm hearing things extremely well and have had some very experienced engineers and producers come help me with speaker placement, etc..... and they have been impressed with the sound/room in general, including the stereo field.

My question is, given the above, do you feel I am sacrificing some quality in not being setup with a totally passive volume control for the mains? Serious question...... I had as I said earlier, considered the passive setup, in that the same guys mentioned above also use passive means for their mains. I'll feel a little more comfortable having something to grab volume wise in an emergency, and passive is more apart from the computer than a digital controller for the RME master buss volumes.

Thanks.

kdm

07-18-2009, 01:03 AM

Whatever your needs are, whether you mix ITB, through a summing mixer or a hi-end analog mixer, the number of converters you need buy has to do with the number of different analog streams you play at the same time.

So for example if you only mix stereo ITB, all you need is a single stereo DA converter.

If you mix stereo AND record up to 2 channels at a time you need to add a stereo AD converter.

If you record up to 2 channels at a time, monitor in stereo and feed a stereo headphone mix you need 2 channels of ADC and 4 channels of DAC, and so on and so forth.

Yes, we all know this. You could simply have said - "accomodating additional switchable monitors requires 4 more DA with the control room solution vs. the passive solution, adding in the cost of either a switchable passive control or a separate switcher.

The number of I/O for main monitor, submix outs and record is the same either way. Only switching between different monitors changes the count, but really only slightly by comparison - even switching between 3 stereo monitors only increases the output count by 4. Considering the number of 8 channel I/Os already involved and studio expense in general, accounting for 4 extra ADs isn't a big deal.

My setup is 5.1 with a secondary stereo monitor system, switchable, and covered with I/O already. You still don't seem to understand that I have no problem with you claiming that you prefer passive and why, but the "pros only" condescending approach isn't supporting your expertise.

These additional converters cost MONEY + you HAVE to use digital volume controls that reduce bit depth AND produce digital errors in a critical listening environment, so while it may appeal to a musician or the recording enthusiast, Steiny's CR is useless to a pro.
It really isn't very professional to insult people just to make yourself appear more knowledgeable or experienced. You could simply say, "I disagree and here's why".

Funny, I didn't see you react to personal attacks all along this thread which have nothing to do with 'balanced or accurate approach', it's just MY approach that seems to bother you.
I never personally attacked you, only challenged your approach here. You are the only one that singled out what you considered "pro".

So I guess that for you personal attacks in a technical debate are not 'defensive, opinionated BS approach' (No offense).

But your selective preference regarding one approach that by a strange coincidence happens be the opposite view, smells like defensive, opinionated BS.

No offense.Come up with your own original argument rather than paraphrasing mine.

I said specifically "use what works for you" several times specifically to give other opinions credit where credit is due. You are the only one pushing your opinion on everyone else with arrogant and condescending "pros only use..." remarks.

Good luck with your ventures RSE. Sounds like you have a successful career. Try a little harder to reflect that with a bit more class. Sorry, but I'm out of time for this.

RSE

07-18-2009, 03:47 AM

RSE..........

Look, I'm just a composer who gets by on as much tech info as needed to get the job done...............

I have a machine room and to go passive on the volume control, I'm looking as about a 6' run to each powered amp.

-SNIP-

My question is, given the above, do you feel I am sacrificing some quality in not being setup with a totally passive volume control for the mains?

Hi paulwr,

To answer your question, you are sacrificing some quality at the moment, but mainly because you use digital volume controls.

There are two issues here: using a digital monitor volume control, and the active vs. passive question.

The first issue is very clear: active or passive, an analog volume control is better then what you are using right now for the reasons mentioned earlier.

The second question is trickier, because it has to do with electrical data that is unknown to you at the moment, so here are the guidelines:

Passive volume controls are inherently better of course, since you avoid the distortion, noise and coloration of an additional amp.

BUT OTOH:

Cables are in fact capacitors, because they are made of two (or more) conducting leads that are physically very close to each other, just like the capacitors in the electrical circuits in your studio.

Capacitors can be used to filter out high frequencies and that's a problem: the whole chain after the passive volume control's output acts as a low pass filter.

There's nothing wrong with low pass filters in the signal's path, as long as the filtered frequencies are higher then the frequencies we need to hear - the smaller the total loading capacitance on the volume control's output, the higher the LPF frequency.

'The total loading capacitance on the volume control's output' is the total of:

1. The capacitance of the volume control's output (very small)
2. The capacitance of the power amplifier's input
3. The capacitance of the cable - which is usually the dominating factor.

The longer the cable > the higher the capacitance > the lower the LPF frequency -

This is why we use the shortest cables possible, and choose cables which are manufactured to have as low capacitance as possible.

So I can't answer this question without the relevant data, thus here are your options:

1. You can figure out your amplifiers' input capacitance + your cables capacitance and calculate the total loading capacitance on a volume control's output in your particular case, and so figure out the LPF frequency

2. You can hire in/loan demo units of passive (~ 10KOhm) and active volume controls and listen for yourself, which IMO is always the better option.

In any case you need to make sure that your cables are as short as possible and with the lowest capacitance possible - check out the manufacturers' sites, they provide the relevant data for their various cables.

---

In a way, I'm using passive already, because my Event Precision 8's have an input pad, which I have adjusted to keep my RME control pannel volume control at -20 to -9 db for listening at various volumes, though I generally keep at about an average of that and monitor at moderate levels.

With digital faders this may be better because you are using a higher bit depth in your converters, but once you get an analog volume control you may find that your speakers sound better with the attenuators off (your converters will then be streaming the maximum bit depth, of course).

In my particular case the ADAM S3As sound better with their input gain control cranked up to the maximum, and the 77dBSPL point is at ~-30dB on the NHT.

I hope this helps,

RSE

paulwr

07-18-2009, 04:50 AM

I'll give this a shot. A good friend is a dealer for Hosa and I'll start with their passive "Hosa mvs-413" first, and some shorter low capacitance cables. Helps that I get it at dealer cost. I look forward to a/b ing with what I do now, and from other info I've gathered from this thread, I should probably check the passive volume against the digital down the road in case anything is degrading over time.

I'll check with different positions on the pads at the speaker amps as well. Makes sense to leave them up and rely on just one volume control to me, though.

This has been a very informative thread, at least for me, all the way around.

Thanks

RSE

07-18-2009, 06:35 AM

The number of I/O for main monitor, submix outs and record is the same either way. Only switching between different monitors changes the count, but really only slightly by comparison - even switching between 3 stereo monitors only increases the output count by 4. Considering the number of 8 channel I/Os already involved and studio expense in general, accounting for 4 extra ADs isn't a big deal.

Voila, you ASSUME that everyone needs what YOU think is the optimum number:

"... the number of 8 channel I/Os already involved and studio expense in general..."

But a pro mixer mixing in stereo ITB (not recording anything) needs only 2 channels of DAC, and a pro mixer needs a large number of monitor options - I use 5, for example

A pro mixer (not recording anything) using a 16 input summing box needs ~16 DACs, he may use 14 for mixing, 2 for monitor. If he uses analog gear he may get 8 extra DACs and use 16 for summing, 2 for monitor, and 6 for his analog inserts / sends.

For using Stieny's CR according to your formula, they all need to buy an extra 8 channel DAC - 1700-2000 $US in Apogeeland.

All studios using large format analog desks need AD/DACs in groups of 8 or 16, and anyway don't need the Steiny CR since such analog desks have their own CR section.

For all of them, Steiny's CR is useless.

Nevermind, let's check your formula anyway:

Let's say that the happy studio owner bough an 8 DACs unit for his Steiny CR, and uses 3 sets of monitors as you suggested:

So 6 DACs are used for monitoring, the last 2 for the headphone mix.

Now if the poor bastard craves for any analog piece of gear - a master tape machine, a reverb, a compressor, whatever - he needs to buy an extra 8 DAC unit.

If he wants a second h/p send, or add (God forbid) a forth monitor option - he needs to buy an additional 8 DAC unit.

You think that '4 extra ADs isn't a big deal' -

4 extra DACs is HALF THE TOTAL NUMBER OF DACs in that '8 channel I/Os already involved and studio expense in general'...

Since when designing a general CR solution the final number of the client's DACs is unknown -
then 'in general' 4 extra DACs should be calculated as half the price of an 8 DAC unit - that's 1700-2K$US in Apogee money.

You ASSUME that everyone needs what YOU think is the optimum number -

To quote YOUR OWN words: "It really isn't very professional to insult people just to make yourself appear more knowledgeable or experienced" :D

LEX

07-18-2009, 06:48 AM

Alright.

Let me put this out there, and whoever tell me my "chain" is wrong.
I have a full MOTU 2408 424 system.

For my digital gear, I have an RME ADI 648(sic), that is hosted via light pipe to AES.

MY output setting from Nuendo is out 15-16 (bank b 4), part of the Digital I/O, 7-8 goto an AES to SPDIF converter into the Presonus Central Station via SPDIF.

I'll bet you my room is more accurate than anyone's here.

Either way, what I am listening to and what they are listen to are 2 different things, and are going to sound completely different.

Whether I need 2 channels or 5.1, PT's I cut, edit, review and deliver. It hasn't failed yet.
Nuendo has failed every time, but that isn't exactly news.

SB = Midi
Anything but SB = audio.

LEX

RSE

07-18-2009, 07:05 AM

Referring to other professionals as less than such is a general insult, as opposed to a personal one. Is one "morally superior" to the other?

:pop_corn:

That's exactly the kind of BS that turned formerly serious and informative forums like Gearslutz to what they are today.

This politically correct approach drove out many pros from internet forums, since they had to put up with all the pro-wannabe BS - people who dubbed themselves 'pro' just because they bought some expensive gear, or hobbyists that insisted on carrying long ignorant debates because 'on the internet everyone's equal'.

...and we all missed on a great opportunity to learn from these pro's experience.

I don't know most of the people who post over the net - I know that most posters are NOT professionals, so this is my initial assumption - even if they have a fancy web site, expensive gear or a suspicious looking credit list.

I recognize other pros by what they post, by knowing their work or by knowing them in person - and only then I address them as 'pros' -

I expect others to treat me the same way, so taking such approach as an insult only implies that the insulted person is not a pro :wink:

A genuine professional doesn't need the recognition of an obscure forum poster, and that's why I don't mind the personal attacks -

When I'm not there, you may even beat me up for all I care :D

RSE

07-18-2009, 07:12 AM

I'll give this a shot.

-SNIP-

I should probably check the passive volume against the digital down the road in case anything is degrading over time.

Good luck, and IMO check the active volume control as well - whatever works best.

RSE

colony nofi

07-18-2009, 07:26 AM

RSE!
hey mate - I'm Brendan Woithe - don't think I've caught your name around here yet?
I think I might be opening up an entire other topic, but I just wish to explore part of your argument, and see if we can collectively come to some conclusions.

Its the argument about digital faders, and perhaps then the cr in general.

I will preface this by saying I run loads of metric halo 2882's, and until recently have run my monitoring through an spl 2381 monitor controller prior to going to powered monitors.

Lets have a closer look at what it means to lower a signal digitally...

Lets ignore the fact that nuendo is a 32bit float system, and for ease just treat our little system as a fixed point 24bit system - both the mix engine AND convertors.

Our mix is happily done, peaks just below 0dBFS, and goes out to our convertors at that level, and comes out as a line level signal. We are using the full 24bits of information inside the DAW, while still understanding that the last 2 LSB will always be noise....
So what happens to our audio signal when we reduce our monitoring fader by say 24dB?
We reduce the dynamic range of that audio by 24dB. (approx 6bits....)
Have we added any distortion to the signal? NO.
We have simply reduced the dynamic range of the signal by 24dB...

I will repeat that again because it is important.

We have not introduced ANY extra distortion to the signal. We have only "added" more random noise.

So now then I ask you. What is the problem with this?

I'm interested to know.

Brendan.

07-18-2009, 09:19 AM

That's exactly the kind of BS that turned formerly serious and informative forums like Gearslutz to what they are today.

This politically correct approach drove out many pros from internet forums,

It was a question. I am not politically correct. I understand the problem. It has always been my contention that the internet has created an age of enlightened stupidity. Anyone can find the answer. Few know how answers are derived, yet the knowledge of how an answer is derived is crucial to understanding any subject.

I don't use the "Control Room". I haven't given SB a dime since they cut off their promised update to SX3. I have only a single set of monitors. I am not a pro, unless the quality of my output defines me as such.

Carry on.

P.S. I presume I must know who you are from somewhere else, since you said I am "a talented musician". Few people would bother to look up someone's music just because they post in a forum. Of course, the proper response to such a declaration would be, "thank you".

The Guru

07-18-2009, 09:45 AM

To me, the art is translating what you hear from whatever you use to monitor while mixing to what sounds good across a broad spectrum of "user" devices. To that effect, it doesn't matter what tools you use, just how well you translate what you hear to what you want to hear.

Why worry about all of the above when the end playback device is an MP3 player?

TAFKAT

07-18-2009, 11:43 AM

A genuine professional doesn't need the recognition of an obscure forum poster, and that's why I don't mind the personal attacks -

:icon_rolleyes:

So let me be clear on this, you are a genuine professional, while others here posing a perspective or opinion that differs, of course in your opinion, are not.. ? !!

Seriously Mate, you have you head so far up, its amazing you can breath !!

IMO, a genuine professional does not need to hide behind cyber anonymity, displaying the tact that you have displayed. None of us know you from a bar of soap past what you have posted here , but you landed here guns blazing as if you are some omniscient authority on all that is that we should all stop in awe of, with an attitude not of contributing , but dictating, berating and personally attacking members. Members that have contributed hugely to the knowledge base here and that I personally have the utmost mutual respect for, who show consummate professionalism not only in their approach towards others opinions, but also their overall manner. You on the other hand just remind me of some self anointed Gearslutz drone that believes that because you have some experience, that makes you an authority over everyone else..

Personally I could care less what you have to offer from here on in as I can't get past your arrogance, and I personally have no time for cyber anonymous super heros who expect everyone to fall at their feet after some long winded resume recital , as if that allows them any more weight.

Shame really, I am sure you do have plenty that you can contribute, I just can't get anything past your S/N , but carry on , I am sure you like nothing better than to listen to your own voice.. :sleeping:

P.S Notice no capitals, as soon as someone types anything in capitals to highlight a point, to me that indicates they need to shout to try and make that point , and if they need to shout , then they have already lost..

kdm

07-18-2009, 02:21 PM

Voila, you ASSUME that everyone needs what YOU think is the optimum number:

"... the number of 8 channel I/Os already involved and studio expense in general..."

A pro mixer (not recording anything) using a 16 input summing box needs ~16 DACs, he may use 14 for mixing, 2 for monitor. If he uses analog gear he may get 8 extra DACs and use 16 for summing, 2 for monitor, and 6 for his analog inserts / sends.

For using Stieny's CR according to your formula, they all need to buy an extra 8 channel DAC - 1700-2000 $US in Apogeeland.

RSE, I really think you aren't reading or understanding what I'm writing. You keep turning my words around to change your argument - makes no sense.

I never assumed anything - I was agreeing with you that you need extra DA for additional monitors, but seriously man, $1200 for converters is a big deal to a successful studio???

Sorry, but that just isn't a big deal vs. the cost of running a studio, esp. considering the Avocet (my next 5.1 monitor controller) is going to run me $3000. Dolby LM100 is $3000 just broadcast/network standard meters, (mainly because its' Dolby). Surround monitoring is $3k-$15k for a small to mid sized facility. ProTools HD 3/4 for an *additional* edit/mix room for small projects (e.g. not feature film) - $15-20k. MC Pro just for an editor - $18k with Eucon (Eucon is $2500 for Nuendo).

Seriously, $1200 for DA just isn't a big deal in that context. My next string library will cost that, and I already have 10x that into sample libraries alone. My next DA will cost twice that (Mytek, Cranesong, etc).

That's my point in challenging your cost analysis - justifying a $120 PVC plus whatever you put into a switcher ($500-$1000?) vs. a $1200-1500 DA is nit picking costs in the midst of an already massive investment that I'm sure you have incurred as well. There are better reasons to argue for whatever config works for you - as in, "this works best for me".

All studios using large format analog desks need AD/DACs in groups of 8 or 16, and anyway don't need the Steiny CR since such analog desks have their own CR section.

*All* studios don't use large format analog desks. You aren't in any of the various areas of post, scoring, sound design, pre-production, etc. Large and mid-sized post dub/mix stages use large desks and multiple PT rigs as others here can attest - usually larger in channel requirements and floor space for the mix room than the type of studio you are describing.

Sound designers and composers don't use large format analog consoles - most of us are all-digital, or mostly digital. Editors might have an MC Pro with ProTools, Nuendo, Pyramix (isn't Alan Meyerson using Pyramix and/or PT now Lex? He was on Nuendo) etc.

For all of them, Steiny's CR is useless.

Maybe for all of the studios you know, but not for all studios necessarily. You are simply extrapolating your own experience recording bands onto every other studio/pro in the audio world. It's only true of studios *like* yours. Not of other professionals in the music/audio industry.

*** But to clarify before I get blasted by the digital police since I've been a big advocate of accuracy in understanding digital audio for years - I'm not defending the control room as if it's the best thing since sliced bread - it's fine for what it does, but I happen to agree that it isn't the best solution most of the time - logistically mainly. Certainly not useful for recording bands, ensembles, orchestras, etc. No software monitoring solution works there.

Loss of bit depth *is* a concern, but, *it isn't the only one* or the most important since it's there in every mix where you use faders in your DAW, esp. in ProTools.

And as Brendan pointed out, it's a loss of dynamic range first and as I've pointed out, there is a loss in *resolution* which is more significant than quantization error (though they are related to a degree). The quantization error issue is there regardless of whether you drop the faders or not - it's a simple fact of digital audio - dropping the main fader only moves where in the amplitude of that material it happens.

Sorry man, but that was my major and career field before going full time into music/audio years ago.

I've said how and why I agree with you in other parts of this thread, and you still came on the attack. To me, that's not a professional attitude, regardless of your experience and credit list - it simply undermines your credibility.

----
Sorry to have been a bit deceptive here, but my only point in responding to you here was to see if you would take a balanced view after being challenged on your rather aggressive approach. That would be a no. Question answered. :wink:

I don't do this often (take a side to argue just to challenge another poster), but your "pros only" reference in your first comment on this thread and response to Sam and others wasn't true or fair since I know that these guys here are in fact quite pro and have some impressive experience and expertise.

It's also not how I respond to non-pros and amateurs, or what I expect of other professionals I work with.

Regardless of how successful any of us is or isn't, I see no need for arrogance. Forums are a great place to learn and help one another, and inevitably piss people off as well, but the "pros only" mindset simply sounds like a desperate attempt to validate oneself.

@ Lex - sorry bro - I hope you didn't think I was bashing your choice of the Presonus in your studio (the one of them at least) - sorry if it sounded that way. I know your high level of work and experience; and your choice of gear is based on your skill, knowledge and needs of the given studio you are working in, not a singular view of traditional studio configurations.

RSE

07-18-2009, 05:09 PM

RSE!

Its the argument about digital faders, and perhaps then the cr in general.

-SNIP-

Lets have a closer look at what it means to lower a signal digitally...

Lets ignore the fact that nuendo is a 32bit float system, and for ease just treat our little system as a fixed point 24bit system - both the mix engine AND convertors.

Hi Brendan,

A fixed point 24 bit mix engine is not a good idea, because if a 0dBFS recorded signal's level is raised you have no bits left, and if it's lowered bits get truncated. Furthermore, when mixing signals together the overall level will rise, and then all the faders need to be lowered and the individual channels' bits get truncated again.

That's the reason for the need of more bits at the mix buss - a typical example of a decent mix buss in a fixed point system is ProTools' 48 bit mix buss.

---

Our mix is happily done, peaks just below 0dBFS, and goes out to our convertors at that level, and comes out as a line level signal. We are using the full 24bits of information inside the DAW, while still understanding that the last 2 LSB will always be noise....
So what happens to our audio signal when we reduce our monitoring fader by say 24dB?
We reduce the dynamic range of that audio by 24dB. (approx 6bits....)
Have we added any distortion to the signal? NO.
We have simply reduced the dynamic range of the signal by 24dB...

I will repeat that again because it is important.

We have not introduced ANY extra distortion to the signal. We have only "added" more random noise.

So now then I ask you. What is the problem with this?

I'm interested to know.

Brendan.

This time I'll respectfully quote from a Bob Katz's article (http://www.digido.com/articles-demos.html), he explains it with a very clear analogy:

Follow that Sample

Let's start with a little lesson in DSP (Digital Signal Processors). Many workstation and processor manufacturers ignore the critical issue of wordlength. Let's examine what happens to digital audio when you change gain (or mix, equalize, compress, sample rate convert, or perform any type of calculation) in a digital audio workstation. It's all arithmetic, isn't it? Yes, but the accuracy of that arithmetic, and how you (or the workstation) deal with the arithmetic product, can make the difference between pure-sounding digital audio or digital sand paper.

All DSPs deal with digital audio on a sample by sample basis. At 44.1kHz, there are 44,100 samples in a second (88,200 stereo samples). When changing gain, the DSP looks at the first sample, performs a multiplication, spits out a new number, and then moves on to the next sample. It's that simple.

Instead of losing you with esoteric concepts like two's complement notation, fixed vs. floating point, and other digital details, I'm going to talk about digital dollars. Suppose that the value of your first digital audio sample was expressed in dollars instead of volts, for example, one dollar and fifty one cents--$1.51. And suppose you wanted to take it down (attenuate it) by 6 dB. If you do this wrong, you'll lose more than money, by the way. 6 dB is half the original value (it has to do with logarithms; don't worry about it). So, to attenuate our $1.51 sample, we divide it by 2.

Oops! $1.51 divided by 2 equals 75-1/2 cents, or .755. So, we've just gained an extra decimal place. What should we do with it, anyway? It turns out that dealing with extra places is what good digital audio is all about. If we just drop the extra five, we've theoretically only lost half a penny--but you have to realize that half a penny contains a great deal of the natural ambience, reverberation, decay, warmth, and stereo separation that was present in the original $1.51 sample! Lose the half penny, and there goes your sound. The dilemma of digital audio is that most calculations result in a longer wordlength than you started with. Getting more decimal places in our digital dollars is analogous to having more bits in our digital words. When a gain calculation is performed, the wordlength can increase infinitely, depending on the precision we use in the calculation. A 1 dB gain boost involves multiplying by 1.122018454 (to 9 place accuracy). Multiply $1.51 by 1.122018454, and you get $1.694247866 (try it on your calculator). Every extra decimal place may seem insignificant to you, until you realize that DSPs require repeated calculations to perform filtering, equalization, and compression. 1 dB up here, 1 dB down here, up and down a few times, and the end number may not resemble the right product at all, unless adequate precision is maintained. Remember, the more precision, the cleaner your digital audio will sound in the end (up to a reasonable limit).

The First Secret of Digital Audio

Now you know the first critical secret of digital audio: wordlengths expand. If this concept is so simple, why is it disregarded by some manufacturers? The answer is in your wallet. While DSPs are capable of performing double and triple precision arithmetic (all you have to do is store intermediate products in temporary storage registers), it slows them down, and complicates the whole process. It's a hard choice, entirely up to the DSP programmer/processor designer, who's been put under the gun by management to fit more program features into less space, for less money. Questions of sound quality and quantization distortion can become moot compared to the selling price.

Inside a digital mixing console (or workstation), the mix buss must be much longer than 16 bits, because adding two (or more) 16-bit samples together and multiplying by a coefficient (the level of the master fader is one such coefficient) can result in a 32-bit (or larger) sample, with every little bit significant. Since the AES/EBU standard can carry up to 24-bits, it is practical to take the 32-bit word, round it down to 24 bits, then send the result to the outside world, which could be a 24-bit storage device (oranother processor). The next processor in line may have an internal wordlength of 32 or more bits, but before output it must round the precision back to 24 bits. The result is a slowly cumulating error in the least significant bit(s) from process to process. Fortunately, the least significant bit of a 24-bit word is 144 dB down, and most sane people recognize that degree of error to be inaudible.

Something For Nothing?

But suppose you want to record the digital console's output to a 16 bit medium, like the CD. Frankly, it's a serious compromise to take your console's 24-bit output word and truncate it to 16 bits. After processing, the mastering engineer uses a technique called dithering to take long wordlengths, and cleanly turn them to 16-bit for the CD. First, must ensure that our DAW is high resolution (has very low distortion at low levels) and can be bit-transparent when called upon. Bit-transparent means that the output is identical to the source, from the most significant to the least significant bit, that the DAW does not increase or decrease the source wordlength.

Good Advice

Once you've verified your workstation is bit-transparent, then proceed with editing, with the goal of maintaining the integrity of your original source. Do not change gain unless you need to align the gains of two pieces you are editing together. Do not normalize (normalization is just changing gain). Do not equalize. Do not fade in or fade out. Just edit. This is to avoid additional DSP or degradation when the mix gets to the mastering studio .Leave the segues, fadeouts and gain changes for the mastering house, where they can properly handle the long wordlengths necessary for smooth fades (so that's why your last fadeout sounded like it dropped off a cliff!). Follow these simple guidelines and your digital audio will immediately start sounding better.

Part II

Dither

-SNIP-

Some Tests for Linearity

You can verify whether your digital audio workstation truncates digital words or does other nasty things, without any measurement instruments except your ears. Obtain the disc Best of Chesky Classics and Jazz and Audiophile Test Disc, Vol. III, Chesky JD111.* Track 42 is a fade to noise without dither, demonstrating quantization distortion and loss of resolution. Track 43 is a fade to noise with white noise dither, and track 44 uses noise-shaped dither (to be explained). Use Track 43 as your test source; you should be able to hear smooth and distortion-free signal down to about -115 dB. Then listen to track 44 to see how much better it can sound. Try processing track 43 with digital equalization or level changes (both gain and attenuation, with and without dither, if it's available in your workstation) to see what they do to the sound. If your workstation is not up to par, you'll be shocked. Use a quiet, high-gain headphone amplifier to help reveal the low level problems.

*available at major record chains or through Chesky Records, Box 1268, Radio City Station, New York, NY 10101; 212-586-7799. The hard-to-find CBS CD-1, track 20, also contains a fade to noise test.

So Little Noise, So Much Effect

-96 dB seems like so little noise. But strangely, engineers have been able to hear the effect of the dither noise, even at normal listening levels. Dither noise helps us recover ambience, but conversely it also obscures the same ambience we've been trying to recover! Dither noise adds a slight veil to the sound. That's why I say, dither, you can't live with it, and you can't live without it.

-SNIP-

--------------------------------------------------------------------------------
Copyright Digital Domain, Inc. We invite you to link to our site, which will be periodically revised.

RSE

07-18-2009, 06:42 PM

I never assumed anything - I was agreeing with you that you need extra DA for additional monitors, but seriously man, $1200 for converters is a big deal to a successful studio???

Sorry, but that just isn't a big deal vs. the cost of running a studio, esp. considering the Avocet (my next 5.1 monitor controller) is going to run me $3000. Dolby LM100 is $3000 just broadcast/network standard meters, (mainly because its' Dolby). Surround monitoring is $3k-$15k for a small to mid sized facility. ProTools HD 3/4 for an *additional* edit/mix room for small projects (e.g. not feature film) - $15-20k. MC Pro just for an editor - $18k with Eucon (Eucon is $2500 for Nuendo).

Seriously, $1200 for DA just isn't a big deal in that context. My next string library will cost that, and I already have 10x that into sample libraries alone. My next DA will cost twice that (Mytek, Cranesong, etc).

That's my point in challenging your cost analysis - justifying a $120 PVC plus whatever you put into a switcher ($500-$1000?) vs. a $1200-1500 DA is nit picking costs in the midst of an already massive investment that I'm sure you have incurred as well. There are better reasons to argue for whatever config works for you - as in, "this works best for me".

Your figures are wrong.

The PVC is 79 $US (http://www.listenup.com/NHT+PVC+Pro-p-PVC_Pro-p-50052.html), + the extra switches, box etc. ~120$US = 200 $US

(not 1120 $US).

YMMV with the extra 8 DACs you plan to buy. The cheapest 8 ch Apogee option is 2000 $US.

Your DACs (quote: My next DA will cost twice that (Mytek, Cranesong, etc)) are 2400 $US

200 $US vs. 2000 $US or 2400 $US.

Do the math (again).

---

All studios using large format analog desks need AD/DACs in groups of 8 or 16, and anyway don't need the Steiny CR since such analog desks have their own CR section.

*All* studios don't use large format analog desks. You aren't in any of the various areas of post, scoring, sound design, pre-production, etc. Large and mid-sized post dub/mix stages use large desks and multiple PT rigs as others here can attest - usually larger in channel requirements and floor space for the mix room than the type of studio you are describing.

Sound designers and composers don't use large format analog consoles - most of us are all-digital, or mostly digital. Editors might have an MC Pro with ProTools, Nuendo, Pyramix (isn't Alan Meyerson using Pyramix and/or PT now Lex? He was on Nuendo) etc.

Maybe for all of the studios you know, but not for all studios necessarily. You are simply extrapolating your own experience recording bands onto every other studio/pro in the audio world. It's only true of studios *like* yours. Not of other professionals in the music/audio industry.

Read my quote again:

All studios using large format analog desks need AD/DACs in groups of 8 or 16, and anyway don't need the Steiny CR since such analog desks have their own CR section.

I wrote 'all studios using large format analog desks' not 'all the studios in the world'.

The sentence means that all the studios that use large format analog desks have special needs:

Because most large format analog desks come with 8 channel buckets (= a frame containing 8 channels) they need multiple 8 or 16 ch DAC units to match the total number of channels.

So if they buy an extra DAC for the Steiny CR it will be an extra 8 ch unit - and they don't need Steiny's CR anyway because they got a CR section in the desk..

Again: 'all studios using large format analog desks', not 'all the studios in the world'.

---

*** But to clarify before I get blasted by the digital police since I've been a big advocate of accuracy in understanding digital audio for years - I'm not defending the control room as if it's the best thing since sliced bread - it's fine for what it does, but I happen to agree that it isn't the best solution most of the time - logistically mainly. Certainly not useful for recording bands, ensembles, orchestras, etc. No software monitoring solution works there.

Here's another thought -

Being software based, if Steiny's CR crashes it can send a blast of 0dBFS digital noise to your monitors, frying your speakers and your ears (and your client's ears) with nothing to stop it except the computer's reset button or the converters' mains.

This can take quite some time, if they're in the machine room or a cabinet.

---

Loss of bit depth *is* a concern, but, *it isn't the only one* or the most important since it's there in every mix where you use faders in your DAW, esp. in ProTools.

And as Brendan pointed out, it's a loss of dynamic range first and as I've pointed out, there is a loss in *resolution* which is more significant than quantization error (though they are related to a degree). The quantization error issue is there regardless of whether you drop the faders or not - it's a simple fact of digital audio - dropping the main fader only moves where in the amplitude of that material it happens.

Sorry man, but that was my major and career field before going full time into music/audio years ago.

I've said how and why I agree with you in other parts of this thread, and you still came on the attack. To me, that's not a professional attitude, regardless of your experience and credit list - it simply undermines your credibility.

Please read the article that I quoted in my previous message.

---

Sorry to have been a bit deceptive here, but my only point in responding to you here was to see if you would take a balanced view after being challenged on your rather aggressive approach. That would be a no. Question answered. :wink:

I don't do this often (take a side to argue just to challenge another poster), but your "pros only" reference in your first comment on this thread and response to Sam and others wasn't true or fair since I know that these guys here are in fact quite pro and have some impressive experience and expertise.

It's also not how I respond to non-pros and amateurs, or what I expect of other professionals I work with.

Regardless of how successful any of us is or isn't, I see no need for arrogance. Forums are a great place to learn and help one another, and inevitably piss people off as well, but the "pros only" mindset simply sounds like a desperate attempt to validate oneself.

'The Internet democracy' again. Believe what you want, if it makes you feel better - but 'the Internet democracy' grants me the freedom of opinion as well, and the right to have any 'approach' I choose to have, regardless of your like or dislikes.

You don't like my approach - Ok, got it, and ain't gonna change it.

Now can we move on please?

kdm

07-18-2009, 08:17 PM

Your figures are wrong.

The PVC is 79 $US (http://www.listenup.com/NHT+PVC+Pro-p-PVC_Pro-p-50052.html), + the extra switches, box etc. ~120$US = 200 $US

Discount then - the PVC pro is $119 US everywhere I looked.

200 $US vs. 2000 $US or 2400 $US.

Do the math (again).
You really are looking at this with tunnel vision. I know exactly how to add all of this up and what it costs, as does most anyone looking at their layout and record/monitor needs - I run 48 inputs currently, with up to 16 outs.

You are trying to put a formula on studio costs and apply it to everyone. I explained quite clearly how these costs differ between your studio concept and many others where a simple stereo controller like the PVC isn't even an option. Enough of trying to justify a $120, sorry, $79 controller with studio cost estimates that vary for much bigger reasons than that.

I wrote 'all studios using large format analog desks' not 'all the studios in the world'.
Yes, my apologies - I did misread this one. No need to explain what it means. I know exactly what studios using large format consoles need, having worked in many of them myself as a producer/engineer.

Being software based, if Steiny's CR crashes it can send a blast of 0dBFS digital noise to your monitors, frying your speakers and your ears (and your client's ears) with nothing to stop it except the computer's reset button or the converters' mains.

This can take quite some time, if they're in the machine room or a cabinet.
This is a very good point. Yes, it's a big reason for having *something* between the DAW and monitors - some form of volume control being the best option. But it doesn't even have to be the DAW - a VSTi can do this.

Please read the article that I quoted in my previous message.
I've read quite a bit by Bob Katz (and other digital audio white papers from developers, as well as talking to several in person). The following is a key point in that article:

"-96 dB seems like so little noise. But strangely, engineers have been able to hear the effect of the dither noise, even at normal listening levels. Dither noise helps us recover ambience, but conversely it also obscures the same ambience we've been trying to recover! Dither noise adds a slight veil to the sound. That's why I say, dither, you can't live with it, and you can't live without it."

Allow me to paraphrase in a larger context: the idea of avoiding digital noise can introduce other problems, both within the DAW, and even outside of it when we make other choices in an attempt to avoid those problems. It is important to know what you are trading off before going in as best as possible, but realize that no solution is the absolute best.

As an aside, I've done quite a few tests with dither, comparing Cranesong's analog dither to POW-r, and most of the other popular options. Many are actually quite harsh, and even when added at a low level it can contribute to more of a harsh sound instead of less; or as Bob noted, it can put a thin veil over the sound - quite the opposite of the intended affect, but it depends on the material. Dither isn't a guaranteed improvement. Neither is an analog alternative given phase, crosstalk, increased noise floor, etc. Pointing out the flaws of digital doesn't absolve analog of it's imperfections. Knowing the pros and cons both ways, and making one's own decisions about what are worth worrying about and which aren't is part of engineering, and being taking an individualized approach to the creative process. The latter being the most important element in any studio.

These imperfections on both side are why I said the only "pure" path is all digital to the monitor, with no gain change at any point on any track, and ultimately the highest rate DSD would be optimal, but logistically that isn't realistic for several reasons. And of course this "best path" eliminates 99% of the creative options and reasons for using the gear to begin with. I'll take a minor technical hit somewhere in the chain (even though I prefer not to in general), if it improves my workflow and creativity by 50-100%, or even just 10-20%.

Now can we move on please?Yes, can we? Need I remind you the original topic was N4 vs. PT and why younger kids are adopting it, not RSE's tutorial on "being pro"?

Most of us answered the original question before you posted, and without insulting anyone.

Now, with my mod/admin hat on: Please sideline the attitude from here on out. You might learn something as well, or at least figure out that there are good people with a lot to offer on this forum. And likewise there are many posters that are looking for information you might be able to provide. If you wish to share your experience on issues not related to the marketing of PT vs. N4, feel free to start another thread - a Q&A if you like. Thanks.

LEX

07-18-2009, 08:43 PM

:pop_corn:

RSE

07-19-2009, 03:16 AM

the original topic was N4 vs. PT and why younger kids are adopting it

PT and N4 are different solutions, and PT is much closer then N4 to the 'tactile' concept of the modern large frame analog recording desk.

It took decades of trials and errors and the feedback from thousands of (users) engineers for the analog desk to evolve into its modern tactile form, so we may assume that the current tactile approach of manufacturing recording desks is the most efficient one at this time.

So if we carefully assume that most young kids are into 'building a studio of their own' - ProTools would be more appealing to them then N4, and the natural choice that they would make.

And if cost was a non-issue, that's the choice I would make too:

The reason that the younger guys are moving to PT in droves is because they know that when they get pro they'll eventually buy an expensive system, and far as studio work goes PT is superior, end-of-story.

-SNIP-

In the studio working with live musicians I'll choose PTHD over Nuendo in a heartbeat.

This may explain also why N4 is better then PT for midi work.

LEX

07-19-2009, 06:18 AM

PT and N4 are different solutions, and PT is much closer then N4 to the 'tactile' concept of the modern large frame analog recording desk.

It took decades of trials and errors and the feedback from thousands of (users) engineers for the analog desk to evolve into its modern tactile form, so we may assume that the current tactile approach of manufacturing recording desks is the most efficient one at this time.

So if we carefully assume that most young kids are into 'building a studio of their own' - ProTools would be more appealing to them then N4, and the natural choice that they would make.

And if cost was a non-issue, that's the choice I would make too:

This may explain also why N4 is better then PT for midi work.

Jumping in here. Life raft at the side.

I have to agree with you here.

Being both a Power Nuendo user and a Power PT user, I know the differences.

In terms of being on the traditional side of consoles, PT has it hands down. Just for the fact they have VCA groups.
In ANY mixing situation, VCA group style is key.
Currently Nuendo's grouping really sucks and makes things 4 times harder to automate.
Example, you have a group master, but the minute you want to adjust any channel in that group, the whole group moves.

Bad, poor design. Lack of mixing experience led to that design.

For Post Production work, PT hands down. I don't think I need to go into alot of detail here, but here are just 2 things.

1) 5.1 ITU lock. Not everyone mixes ITU standard. We deliver in many different formats, but prefer mixing in LCR, LSRS, LFE.
Nuendo is locked to ITU.
2) no dual mono support. no 5.1 mono support. Interleave is great, but not when you have to convert your files, all 5 terrabytes of them to interleave just to use Nuendo.
Not to mention, if your 5.1 stems are being conformed, and they are interleaved, you have to break them into mono anyway. There are prelaps ALL the time with these shows I have to add act breaks too.

Interleave is great, but considering the industry, which SB has no clue, mono is what is needed.
Mono 5.1 channels is what most layback houses, and AVID/FCP editors need.

If is is already mono, then there is one less step, one less processed algorhythm that can go wrong.

Kids are building their own studios, but nothing to the extent of some of the power users here.
PT, DP , Logic.

If we are talking music and scoring, it is a different story, though with Nuendo you get video problems. Kind of defeats the purpose.

Midi, Cubendo hands down. I'll stick with Nuendo 3 until I can't anymore.
And SB finally fixes everything after 3 paid upgrades.
Audio, well I am at a loss here.

Reaper, is really really great. But it has its annoying things.
PT, well I still prefer to mix in Nuendo 3 over, even with the crappest automation on the planet.
My system is built around nuendo anyway, so might as well make the best of it.

Reaper, I will try to mix in at some point. This says alot about the constructs of SB software.
Reaper is far more stable, and more reliable than the last 2 versions of SB software.
Once I get used to the workflow, then Nuendo for audio is history.

It is sad because I invested alot into Nuendo. I sucks that people like Fredo and Chris B kept the application from becoming more because they don't understand.

Had high hopes, now it is just reality.
Invest in PT. The return is greater.

LEX

RSE

07-19-2009, 10:21 AM

Reaper, is really really great. But it has its annoying things.

Hi Lex,

I hear a lot of positive opinions about reaper, but haven't had the pleasure yet -

What do you find good (or better then X) in Reaper and what are these annoying things that you've mentioned?

RSE

kdm

07-19-2009, 10:34 AM

Just my experience here on kids getting PT over other apps - most I know of don't know there is a difference between apps, much less what they are. They get PT because it's what they've heard of - every single one I've come across. At one time I consulted to studios with PT rigs, and the knowledge of other apps, and how to use PT was minimal.

Digi truly got the marketing right early on. They have the mindset now that doesn't require any exposure to actual recording to get buyers. That rarely used to be the case. Most new studios started after the musician had some exposure to larger studio recording. Granted, the client lists for most of these new, inexperienced PT studios are usually local bands and local projects. But it's evident in other ways in talking to these guys that marketing sold them rather than knowing why and how to choose one app or another. Of course the stores at least know how to sell PT as being "the app everyone uses", which is all it takes.

I've avoided PT over the years simply on principle of cost vs. capabilities (with some exceptions) and that I do require heavy midi capabilities and despise a multi-app environment (did it with Paris and Logic for years), but eventually I'll at least have to add a PT room if not move all of the post side of my business to PT completely.

Reaper has great potential, but time is still ticking away. If Digi release new HD hardware in the next 6 months, and adds PDC to PTLE, they will seal their dominance for another 5 years at least.

paulwr

07-19-2009, 01:14 PM

Hi Lex,

I hear a lot of positive opinions about reaper, but haven't had the pleasure yet -

What do you find good (or better then X) in Reaper and what are these annoying things that you've mentioned?

RSE

You can download a fully operational current version for free at their website. And the price for a license is very low.

For midi, I don't plan on even trying to figure it out myself. I don't have the time and I know I'll start hearing things if it progresses far enough for me to consider. I make use of the logical editor quite a bit and dont' think I'd be very fast without it.

I'm interested in seeing how it does for mixing........ but really, go try it yourself, it is free...... very quick download and only a couple minutes to install since there is no authorization EVER. And users are paying for it when they find themselves really using it. What a concept.

-Paul

TAFKAT

07-19-2009, 07:28 PM

Kids are building their own studios, but nothing to the extent of some of the power users here.

The point being is why they are going to PT's over the alternatives, and IMO from someone who has been in the frontline supplying DAW solutions for quite a while , the main factors in the initial stages are not based around comparing feature sets, its based more on brand recognition, street credibility, chatter. This is where Digi is king in getting the new blood, they have managed to stamp the "Protools" branding as something that is interchangeable for DAW, as I mentioned earlier.

That is a powerful card to hold, especially now that with the acquisition of Sibelius, they are now snaring the education market that Steinberg had held for many years , so the kids are exposed to PT from the get go. You and I both know that once you are blooded on a DAW, for the most part many don't look elswhere. Case in point for me is that the vast majority of my Steinberg clients are still Steini clients, despite the company being less than stellar. Why, simple, better the devil you know.

Also, for a lot of those clients , PTLE is not an option due to track count limitations, lack of tangible automatic PDC - which is absolutely unforgivable in this day and age , and the lame scalability of RTAS compared to VST/AU. Horses for courses , if the client is requiring some basic tracking, with minimal plugin/VSTi requirements, sure PTLE is fine, but once you break thru the supplied PDC , its a joke having to navigate the ensuing mess.

So for the new blood not focusing purely on tracking bands, but are more focussed on combining creative elements from audio / MIDI / VSTi's, if they would start on Cubendo/SONAR/Logic, etc, PTLE would not even get a look in, simple as that IMO, as it would fall flat on its arse in comparison, yes even PT8..

BTW: Just some clarification on the Automatic PDC on LE, its not that it doesn't have any, as it does , say what.. ?

You read right. Lets think about this , if it didn't have any at all, you would not be able to run any plugins whats ever without it turning into an out of time mess, and that does not happen, and for many who use it for basic minimal tracking and editing, it never even rears its head. I have dozens of PTLE clients who wouldn't even know what PDC is, or that there is a problem.

The PDC in PTLE is limited to the audio buffer setting, so for example if you have the buffer setting set at 256 , you have a total of 256 samples of PDC- across all plugins accumlatively. Now , if you are using plugs with none to very minimal delay- i.e most of the bundled plugs , you can happily track away none the wiser. However start using some 3rd party plugs that have higher inherent delays, and it goes south very quickly. Lower the buffer setting, and of course it heads south way sooner.

For those of us used to Cubendo's/ SONAR's/Logics PDC implementation , where no matter what the buffer setting / track / plugin / VSTi count , its seemless , PTLE is not an option past having a branding in the studio for clients that are none the wiser , but it really isn't a viable option as a replacement , there really is only PTHD , and thats whole other can of worms, as Digi really need to step up and embrace the Native potential of the current technologies , instead of relying on 10 year old DSP Achilles heels to run the plugins. I have no issue with the Mix / Audio engine remaining on the DSP, as that is still PTHD's main strength IMO over Native , but the available DSP for plugins is laughable compared to what is achievable on the current crop of CPU's.

Anyhow,

Thats my 2 cents

:009:

ramagochi

07-21-2009, 01:18 PM

kdm

07-21-2009, 01:32 PM

Thanks for posting the video ramagochi - very interesting!

Animus

07-21-2009, 01:49 PM

nice work ramagochi.

RSE

07-21-2009, 03:37 PM

Wow!! I had work all the weekend and this continues :wink:

I made a video of a passive 50k pot in connected to a Smaartlive in transfer function (Sync pink noise), and I used a very short cable for the reference and a long cable for measure (5m low capacitance and 7m normal capacitance).

http://www.canare.com/ProductItemDisplay.aspx?productItemID=53
http://www.gotham.ch/en/index.php?section=docsys&cmd=64_details&id=5

Sorry for my english :eusa_whistle:

http://www.youtube.com/watch?v=plqA_8N3Joo

The graph are inverted (loss up/gain down)

It looks like you're sending the noise from the (line? headphones?) output, multing one channel through a short blue cable back to the hi-Z input, and the other though the ALPS to the long cable and back to the mic input -

Is this correct?

TerryG

07-21-2009, 04:37 PM

Maybe we can get UAD to model inexpensive passive pot distortion?

But the plugin would cost more than the pot...
:icon_lol:

ramagochi

07-22-2009, 09:10 AM

It looks like you're sending the noise from the (line? headphones?) output, multing one channel through a short blue cable back to the hi-Z input, and the other though the ALPS to the long cable and back to the mic input -

Is this correct?

First I tried to connect one line output to one channel XLR and the other line output to the Hi-Z, I made a transfer function for get the diference between the two inputs, the result was "no diference". Next, I tried to connect one output thru the potentiometer to the "XLR input" (With the wires) and the other output direct to the HiZ, works ok but I had to move the gain (By the Smaart software) every time that I moved the potentiometer.

But, for I made this video, I made a Y cable, and I was connected it by this cable a single output to the two inputs (XLR and HiZ) and I was made a transfer (The blue line on the graph), the transfer for this connection was very good (Flat as you can see in the video). Next, I connected the output of the potentiometer to the two wires, the long cable and the short and I made a transfer.

The transfer with the second system was better (Flatest) than the first option (one channel thru potentiometer and other direct), but good for the example (I didn't need to change the internal Smaart's gain every time that I moved the potentiometer).

I'll trying to get a 10k DACT (I have a 100K unit in a hifi device), and I'll try to make more tests, with phase and distortion.

In other hand, in the passive area, will be better for long cable runs use a commuted attenuation transformer like Sowter (http://www.sowter.co.uk/transformer-attenuators.php) (I want to try one these)

Cheers

P.D. I repeated the test with two diferent sound cards, one Sound Devices USBPre and the Apogee Duet, both had a close results

RSE

07-22-2009, 03:17 PM

Hi ramagochi,

I'm afraid this test's conditions are not 'passive preamp' conditions at all, so I wonder what are you trying to test really.

To emulate 'passive preamp' conditions you need to use a different volume control, avoid using the y-cable, avoid using the Duet (http://www.apogeedigital.com/products/duet.php?section=features)'s mic inputs etc.

Please let us know (as with every experiment) what is your theory, and what findings are you expecting to find.

RSE

kdm

07-22-2009, 03:36 PM

paulwr

07-22-2009, 04:41 PM

Great thread.

The following isn't very scientific, but just as a little more info, I tried turning the pads on my active Event Precision 8's to max pad, and thus upped my RME master fader (all else nominal, such as the Cubase master) by 40+ db over what it was. The sound quality went down. Stereo field narrowed for me and just didn't sound as 'alive'.

I can't seem to find the details about Event's input pad, and it would be good to know what I'm really testing. But here, it sounds BETTER having the master RME volume turned down40-48 db with the input pads on the speakers to max sensitivity (i.e. no padding) than having the pads on to maximum padding and running the RME master up 40-48 db.

I still want to get a passive control for volume so I have something to physically 'grab', esp in an emergency. But I need to know I'm keeping my stereo field, depth, and overall 'life' to the sound that I'm currently enjoying. Hopefully I can borrow one somewhere first; If not I'll just buy one unless the ongoing tests here start seeming conclusive.

If I do stick with digital volume control, I may just get a midi controller for that, and/or a hardware 'kill' switch for
emergencies.

-Paul

kdm

07-22-2009, 05:16 PM

Hi Paul,

I've found the same in the past - pads and generic attenuation as in some monitors just don't provide accurate gain reduction across the frequency range as you would get from digital, or a high quality passive or even active volume control.

The reality is, there will always be *some* tradeoff when putting any attenuation in the audio path. The problem to solve is which is least problematic for a given setup and application.

paulwr

07-22-2009, 07:26 PM

Well, I think my understanding has increased to the point that I'm wanting to come up with stupid solutions........you know, the point that you understand just enough to REALLY embarrass yourself!

Such as: hard wiring the restance needed to just get up to your max volume and no more......... then relying on digital control from that point so that resolution errors are minimal. That way you don't have to worry about the pot degradation, but get some resolution gain from the digital side, since half or more of the level reduction is done passively, but hard wired so no typical pot worries..........

....... feel free to point out just how low I may have sunk here.....

-Paul

RSE

07-22-2009, 07:37 PM

It isn't perfect for a passive gain test in and of itself with the Y and using the Apogee ins (which he said he compared for accuracy first); but it shows the difference the ALPS pot has on frequency response at various gain levels compared to the original transfer function - only at full gain does it approximate the original transfer function's response, though with more loss at the high end - overall it seems rather inaccurate as a gain stage. This is what I expected - most passive gain devices only exhibit accurate response at full gain (e.g. the least amount of resistance).

Hi Kdm,

The loads are wrong, and in such a test one needs to make sure that the sources are capable of driving the loads.

The volume control's value (50KOhm) is way too high - 10K is plenty, the NHT is 4.7K if I remember correctly.

The Y-cable makes the output "see" the ALPS + a HI-Z input (instead of just the ALPS' impedance).

The ref. input is a mic input instead of a power amp input, etc..

All this means that it's not the ALPS that's being measured, but a varying complex system.

If testing the effect of the ALPS in a passive preamp application (that's why I asked, we really don't know what's being tested here), all other conditions should be equal, testing once with the ALPS in the path, and a second time without it.

I.e. the connections should be:

1. Converter line level o/p > short cable (low capacitance) > ALPS (10KOhm or less) > short cable (low capacitance) > optimal power amp > measuring (at least) after the first input stage.

2. Converter line level o/p > short cable (low capacitance) > short cable (low capacitance) > optimal power amp > measuring (at least) after the first input stage.

RSE

kdm

07-22-2009, 07:44 PM

That's basically what a pad is doing - it just cuts the voltage by adding resistance (soaking up power, in a manner of speaking). Also need to account for whether you want balanced or unbalanced, and there are various designs for a pad to consider, but it isn't hard to build one.

RSE

07-22-2009, 07:45 PM

Hi paulwr,

The "pads" in my Adam monitors are active (pots) yet still they react in the same way...

kdm

07-22-2009, 07:53 PM

paulwr

07-22-2009, 08:37 PM

RSE had posted earlier that he did not get the best sound using any attenuation on the Adams. They they were best turned to max sensitivity (no pad). Couple of pages ago.... in answering a question from me....

Just no substitute for A/B'ing this stuff with ears. I haven't met a music consumer yet doing his listening with measuring devices. Perform/mix/master with ears, they listen with ears.

-Paul

RSE

07-23-2009, 12:04 AM

Hi RSE - that's the variable I was wondering about.

I agree with your connection suggestion for a more accurate test.

Do you get a good response with attenuation in your Adams? That's valuable info to consider for anyone looking for monitors.

RSE had posted earlier that he did not get the best sound using any attenuation on the Adams. They they were best turned to max sensitivity (no pad). Couple of pages ago.... in answering a question from me....

True, and now when I think about it, this way I also get to use a smaller (physical) range on the NHT volume control - which makes it even more consistent.

To answer your question the ADAM S3As are amazing monitors, but in less-then-ideal conditions they are more sensitive to room acoustics then any other midfield/nearfield monitor system I know - setting them up in my room was a bitch.

---

Just no substitute for A/B'ing this stuff with ears. I haven't met a music consumer yet doing his listening with measuring devices. Perform/mix/master with ears, they listen with ears.

-Paul

This is very true, the ears are more sensitive then any measuring device.

However when hearing a problem, measuring helps to pinpoint the cause.

RSE

paulwr

07-23-2009, 01:27 AM

True, and now when I think about it, this way I also get to use a smaller (physical) range on the NHT volume control - which makes it even more consistent.

To answer your question the ADAM S3As are amazing monitors, but in less-then-ideal conditions they are more sensitive to room acoustics then any other midfield/nearfield monitor system I know - setting them up in my room was a bitch.

---

This is very true, the ears are more sensitive then any measuring device.

However when hearing a problem, measuring helps to pinpoint the cause.

RSE

Since you have the NHT, and we're speaking of ears...... can you tell me the difference you hear using the NHT for attenuation in the ranges you most listen in.... approx db estimate......... against the NHT being all the way up, and the same db approx range being created with your audio device master volume or daw master volume control?

Thanks,
-Paul

RSE

07-23-2009, 04:35 AM

Since you have the NHT, and we're speaking of ears...... can you tell me the difference you hear using the NHT for attenuation in the ranges you most listen in.... approx db estimate......... against the NHT being all the way up, and the same db approx range being created with your audio device master volume or daw master volume control?

Thanks,
-Paul

Hi Paul,

I mix and monitor at 77dBSPL, NHT @ ~30dB.

For your test I listened to Incubus' 'Love Hurts', it's quite loud so I set the NHT @ ~-40dB.

When I compared the levels with the NHT @0dB, the N4 master was set @ -41.5 dB.

The first thing I noticed was the noise - I monitor through the Neve 8816 mixer and the S3As (http://www.adam-audio.de/studio/midfield/s3a_data.htm) are 360WRMS (just hiss and no hum, but unacceptable in a studio environment).

The image was smeared, LF punch was lost, the lead vocal was coming from anywhere but the front, and the HF was (bigtime) harsh - as if the song was recorded using only cheap chinese condenser mics.

Hope it helps,

RSE

RSE

07-23-2009, 04:52 AM

P.S:

I repeated the test bypassing the Neve -

The s/r ratio was much better, and it sounded not as harsh - but the loss of punch and extra (super) HF were still there.

ramagochi

07-23-2009, 06:12 AM

Hi RSE - I could be wrong as I couldn't trace the connections in the video, but it appeared to be a simple comparative test - transfer function raw into Smartlive (top signal) vs. the split of that signal via the ALPS pot via the other channel into Smartlive (lower signal)

It isn't perfect for a passive gain test in and of itself with the Y and using the Apogee ins (which he said he compared for accuracy first); but it shows the difference the ALPS pot has on frequency response at various gain levels compared to the original transfer function - only at full gain does it approximate the original transfer function's response, though with more loss at the high end - overall it seems rather inaccurate as a gain stage. This is what I expected - most passive gain devices only exhibit accurate response at full gain (e.g. the least amount of resistance).

I also see differences between the two cables, though that should probably be separated from the rest of the test.

I could be wrong, but that's what I interpreted here. Ramagochi might have a better explanation.

Hi kdm,

I did the test only for show whats happen with the signal when pass thru a potenciometer with a long run of cable.

The video test isn't perfect but is for made for an explanation, first I made an accurate test: one channel pass thru the potentiometer and the reference not (Signal on the XLR and test on HiZ on Duet, and signal and reference on the line in of the USBPre (I used a Benchmark DAC1 for a Balanced output)), I taken a transfer of the system for comparision, this is enougth for did this test, but need to correct the gain every time that I moved the potentiometer.

In the video the signal charges in the two inputs at the same time, this is better for the show what happends but the graphs are better than the separate test. I taken a reference graph with short cables, and without potentiometer as well (Using the same box), you can see the result in the blue graph (As well you can see the coherence graph of this test for get a idea of the accuracy)

I only had a 50kHz potentiometer for did the job, and this is very good for the HiZ input, with a 10K pot you'll get a marginally better graphs, but not good.

Cheers

ramagochi

07-23-2009, 06:26 AM

1. Converter line level o/p > short cable (low capacitance) > ALPS (10KOhm or less) > short cable (low capacitance) > optimal power amp > measuring (at least) after the first input stage.

2. Converter line level o/p > short cable (low capacitance) > short cable (low capacitance) > optimal power amp > measuring (at least) after the first input stage.

RSE

I don't think that you need a power amp, any signal input are enought for do the job.

My first test was:

Channel One: Converter line output > short cable 20cm (low capacitance) > 50k Alps > 50cm cable (Low capacitance)/5m cable (Low capacitance)/7m cable (Regular capacitance)> line input of a sound card (Duet or USBPre).

Channel Two: Converter line output > short cable 20cm (low capacitance) > line input of a sound card (Duet or USBPre).

With a transfer function analyzer (Or dual FFT) you don't need make twice the analysis, you'll compares one channel against the other channel, for an accurate measuring you only need two identical channels, and you can test the similarity of these channels with a simple transfer between channels.

The video measure was better than the measure with this system.

Cheers

RSE

07-23-2009, 06:53 AM

I don't think that you need a power amp, any signal input are enought for do the job.

No, different inputs have different conductance and impedance values - that is why we can (for example) use 40 feet mic cables, while line cables need to be short.

The 'passive preamp' is useless unless certain guidelines are met, I have explained it already in a previous post.

RSE

ramagochi

07-23-2009, 07:13 AM

I did a listening test with this box.

System:

DAW > Benchmark DAC1 > 1m cable > Pass F5 power amp > My monitors .

DAW > Benchmark DAC1 > 1m cable > 50k Alps > 5m cable > Pass F5 power amp > My monitors.

Subjetive mode on With the Alps in comparision with no Alps (Nuendo CR) the sound sounds dull, with a small stereo image, and less air in the highs (My ear isn't very good at very high frequencies, I'm a bit old :-), and less dynamic (specially in the bass section). subjetive mode off

Music:
Stravinsky-L'histoire Du Soldat Conducted by Robert Mandell - Ars Nova (http://www.highdeftapetransfers.com/category/63/Chamber-Music)

YERBA BUENA BOUNCE The Hot Club of San Franciso (http://www.hdtracks.com/index.php?file=catalogdetail&valbum_code=HD030911110925)

My monitors: I use two, amplified and un-amplified. An amplified Dynaudio BM15 and a DIY monitors with the F5 amp (http://www.firstwatt.com/products/f5.htm) the DIY speaker are high sensitivity 95dB 1W/1m, very low distortion 0,1% at 100dB SPL (Two way, 12" low and AMT tweeter)

ramagochi

07-23-2009, 07:14 AM

No, different inputs have different conductance and impedance values - that is why we can (for example) use 40 feet mic cables, while line cables need to be short.

The 'passive preamp' is useless unless certain guidelines are met, I have explained it already in a previous post.

RSE

I don't agree, with a transfer you can see the difference between inputs. In this case the two inputs are identical (Same impedance) (The two XLR inputs of a Duet or the two TRS jack line inputs of a USBPre)

Yes, I read your explanations.

Cheers.

ramagochi

07-23-2009, 08:27 AM

Worse scenario

Like as suggest (Sorry, no amp, IMHO no needed):

Output > short low impedance cable 1m>potentiometer>cable (Short, 5m, 7m), Sound card input (The same channel).

Fuzzmeasure (http://supermegaultragroovy.com/products/FuzzMeasure/) for take the measures

Alps 50K Potenciometer at 12 O'clock (-19 dBu).

Short cable
http://farm3.static.flickr.com/2477/3749195966_42bd303b66.jpg?v=0
5m Canare cable
http://farm4.static.flickr.com/3530/3748407453_8bcabdcb8e.jpg?v=0
7m Gotham cable
http://farm3.static.flickr.com/2457/3748408421_e092b27371.jpg?v=0

Frequency response:
http://farm3.static.flickr.com/2674/3749195600_622689b339.jpg?v=0
Blue = Short
Green= Canare
Red= Gotham
Phase:
http://farm3.static.flickr.com/2527/3749195026_b6b1999093.jpg?v=0
Detail of the HF loss (Superposed graphs)
http://farm3.static.flickr.com/2605/3749296018_12075a5ccb.jpg?v=0

I'll try to repeat with a 10k pot, probably the next week (I'll go to work this weekend)

RSE

07-23-2009, 02:19 PM

I don't agree, with a transfer you can see the difference between inputs. In this case the two inputs are identical (Same impedance) (The two XLR inputs of a Duet or the two TRS jack line inputs of a USBPre)

Yes, I read your explanations.

Cheers.

Sorry that you don't agree, but you are comparing apples to oranges.

There is no 'absolute' frequency response for a cable, it changes depending on the source and load the cable connects.

For example:

Take a 40 ft. cable and connect a studio mic to a desk's mic input, and the frequency response would be acceptable - we do it every day in the studio or in live setups -

Now take the very same cable and connect an electric guitar to a mic input with it - the signal would suffer a major FR loss.

You are trying to deduct from the behavior of a mic input (or line in the other card) about the behavior of a totally different input (a specific power amp, with different capacitance and impedance).

It's the mic/line input in your test and the power-amp-to-be-used input that are not identical - not your twin line/mic inputs.

FR test results may only be referenced to the specific load tested:

In some power amps for example, the first component the ALPS "sees" is a resistor, in others it's a coupling capacitor (that's why hi capacitance power amp inputs often need to be terminated with a 10K resistor for this application) -

This means that even when testing different power amps YMMV (there may be different FRs).

If the video test standards were acceptable, you could connect an electric bass to the ALPS and then through your short/long cable combo to two identical line inputs for example, show some frequency loss and rule out the long cable for microphone applications -

This kind of test means nothing, of course.

paulwr

07-23-2009, 03:16 PM

P.S:

I repeated the test bypassing the Neve -

The s/r ratio was much better, and it sounded not as harsh - but the loss of punch and extra (super) HF were still there.

Thanks for that, RSE. For me here in my studio, my master goes directly out of the RME Fireface 800 to my powered speakers. Even with FF 800 master down at -40 or so db, I get 0 audible noise and a sound that just seems way better than it ought to be....... so if I even maintain what I have using the passive control, I'm going to be happy. If it comes out better, great. I know it should, but we'll see. I wouldn't even be doing anything if I didn't want the out of the box vol control.

Thanks again,
-Paul

paulwr

07-23-2009, 05:27 PM

what would be the main drawback for stepped attenuators? Seems like a big step up from pots at first glance. For just setting volume, seems like .5 or 1 db steps would be fine.

-Paul

RSE

07-23-2009, 11:31 PM

Stepped attenuators are fine, often used in mastering suites.

RSE

07-24-2009, 04:18 AM

Thanks for that, RSE. For me here in my studio, my master goes directly out of the RME Fireface 800 to my powered speakers. Even with FF 800 master down at -40 or so db, I get 0 audible noise and a sound that just seems way better than it ought to be....... so if I even maintain what I have using the passive control, I'm going to be happy. If it comes out better, great. I know it should, but we'll see. I wouldn't even be doing anything if I didn't want the out of the box vol control.

Thanks again,
-Paul

Very good, and after you've tried your options in accordance with the guidelines trust your ears and choose whatever sounds best to you - even if it's not the best theoretical solution:

Many times a technical 'flaw' may compensate for other problems, for example technically 'wrong' speakers compensating for room acoustics (NS-10s in small rooms...).

P.s.: If you're not sure about your Events' input capacitance and your FF800 > passive volume control > Events path is electronically BALANCED (XLR/TRS plugs with 2 leads + shield) - terminate @ the Events' inputs with 10KOhm resistors - i.e.

Pin 2 > 10KOhm resistor > ground (pin 1)
Pin 3 > 10KOhm resistor > ground (pin 1)

You'll lose a fraction of power but you have 40dBs of extra level anyway.

P.s. (2): If you like the RME sound try the RME ADI-8 converters (the DS version for 96Khz) - they sound better then the FF800, and their 25 pin I/O fits the PT HD format - very handy when mixing in PT HD studios that have no external converters.

Good luck,

RSE

efernan

07-25-2009, 04:21 AM

Hey mods,

Wouldn't be nice to split this post in two?

One about the original subject and the other OT about passive volume control, etc. (very interesting endeed, but could be missed because the original post's title).

Maybe the second part (about PVC) could also be moved to the GearLust section.

Cheers!

paulwr

07-25-2009, 09:52 AM

yes, probably should have been split long ago..... but now might be too tough splitting 122 posts.

-Paul

TAFKAT

07-25-2009, 12:30 PM

Hey Efernan,

I was thinking the same thing Mate, just haven't got around to it..

TerryG

07-25-2009, 02:18 PM

I think we could wrap up the relevant PT vs Nuendo segment within the first 11 posts of the thread. :wink:

Then, salvage the passive volume control segment to the appropriate subforum. Soon thereafter, when it's proven beyond doubt to all but one that certain inexpensive passive controls are detrimental to frequency response when attenuated, or camps divide on the merits of passive attenuation versus active or digital, we can bring it back here. :wink:

RSE

07-27-2009, 02:52 PM

Soon thereafter, when it's proven beyond doubt to all but one that certain inexpensive passive controls are detrimental to frequency response when attenuated

1. At least one (my) 'certain inexpensive passive control' was tested in the circuit and the FR came out fine.

2. In a typical digital attenuator example discussed here, 40-48dB of attenuation were needed:

But here, it sounds BETTER having the master RME volume turned down40-48 db with the input pads on the speakers to max sensitivity (i.e. no padding) than having the pads on to maximum padding and running the RME master up 40-48 db.

...so it would be very sad if we find out that 'all but one' of this forum's members don't realize that when attenuating an integer digital stream by 40-48dB, all the acoustical cues and ambience information in the bottom 40-48dB range get wiped out. :D

OTOH, if one can't tell the difference between 24 bit and 18 bit integer, why should 48 missing dBs matter anyway? :wink:

RSE

paulwr

07-27-2009, 03:14 PM

A quality stepped attenuator ala Gold Point is looking pretty good, or if I feel the need for extra controls, the Coleman stepped attenuator looks like a great option. I just don't like the idea of a 'pot' in the middle of my sound, but so far I can't dig up any negatives on the passive stepped attenuator approach. I have a couple of friends that mix a lot in 5.1 going through this same dilemma. Never thought the subject of volume control could be this interesting...... I have the question about it into David Newman's tech assistant as well. I appreciate all the informed comments about this.

-Paul

TAFKAT

07-27-2009, 04:52 PM

I think we could wrap up the relevant PT vs Nuendo segment within the first 11 posts of the thread. :wink:

Hey Terry,

Almost... LOL

I had a read thru and was trying to make a note of how best to separate the commentary as cleanly as possible, and got a severe headache and abandoned the attempt.

From what I can make out the shift of subject matter started with the "Control Room" comment , which unfortunately twisted and turned for a while before settling into the productive signal to noise that we have now.

I think someone who was more intimate with the finer detail will need to do the surgery here , I have basically been a passenger in regards to the finer tech detail and have not been following the thread as closely as others, maybe Dedric would be the best..

The quality of the ongoing commentary shouldn't be buried where it is now..

To the members here actively engaged,

Can someone give me a good title for the thread for the current and ongoing subject matter, and I'll edit and move it as is, and then exorcizing the Nuendo v PT commentary at a later date..

The Guru

07-27-2009, 05:09 PM

Just add "/ Passive Volume Control Discussion".

paulwr

07-27-2009, 06:04 PM

Can someone give me a good title for the thread for the current and ongoing subject matter, and I'll edit and move it as is, and then exorcizing the Nuendo v PT commentary at a later date..

Just save yourself some trouble and post the whole thread in two places with a little explaination at the top of it.

-Paul

TAFKAT

07-27-2009, 06:08 PM

Hey Paul,

I don't mind just moving it, no need to duplicate it.

It would be good to have a title reflecting the commentary tho.

Nates idea was O.K, but I would rather something a little more detailed, maybe " Passive Volume Control - Pros and Cons "

What do you think ?

dcwave

07-27-2009, 06:40 PM

Since I started the thread, I took the liberty of changing the title.

paulwr

07-27-2009, 06:52 PM

yea, thread title looks great to me. This has now started at SoundsOnline forum as well, and I'm putting a link up there since so much has been detailed here so far.

Still wondering: Comments about passive stepped attenuation? This has been looking very good on paper and I'm like to hear any experience with good units out there. It sidesteps the many pot issues. I couldn't find much in the way of bad comments about this approach with quality components.

Thanks,
-Paul

TAFKAT

07-27-2009, 06:55 PM

Since I started the thread, I took the liberty of changing the title.

Thanks Dave... :-)

Now to move it to Gearlust.., I'll do that.

Patanjali

08-10-2009, 08:48 PM

Just some comments about Control Room (CR) and Passive Volume:

1. CR is useful for those with modest control room needs AND who have a single multi-channel soundcard.
A working studio may need more flexibility, more outputs, etc, just like they need more outboard equipment, microphones, etc, to cater for a wider range of recording scenarios.

CR provides a lot of functionality without having to have outboard gear.
Not everyone has a workflow that requires actual physical knobs and buttons, nor the physical handson habits developed over years that may rely on it.
I did not 'grow up' with physical consoles, so they are just awkward for me.
If it comes to the crunch, I could use the set on my m-Audio KeyPro88.

WE only record one vocal or instrument at a time, so there is no need for a sophisticated setup. CR give the flexibility without occupying any more desk space and I have outputs to spare, not being into 5.1 or whatever.

2. I have a relatively cheap passive volume control between the RME FF800 and the Tannoys. I have been caught out in the past when the output of a soundcard has gone stuck at high volume and had no way of tempering it a bit. But I did not want a whole other amplification stage in the main audio path.

When it comes to mixing, then I set it to the standard position. At other, less critical times, it is OK to have less than accurate levels and balance.

evanrabby

09-01-2009, 07:26 AM

use reaper or nuendo 3 with the listen bus, set up startup projects with say 6 post fader aux sends to 6 headphone group outputs, use a behringer bcr2000 if you really think you need knobs, i never have in the studio, albeit i have knobs on my headphone amps.

now as i track, each musician has their own mix, hearing all the nice mixing on the fly and during setup so they get a fully processed mix, looking over to me to adjust anyones level to them only incl reverbs/fx level to their mix only, i get very happy clients, inspired by NON DRY / DULL beautifully processed tracking headphone mixes.

latency of 7 ms total round trip has never been even perceived, except mabe a dj, then i give him a cuemix mix out just for his beats and mix it with the nuendo mix for the rest of the instruments.

no hardware needed. you are all insane to think 24 bit audio inside of the 32 bit floating point engines will render sound quality issues when using attenuation. ill have to dig up my article link on a blind test on passive vs digital summing just to give a related reference. it basically embarassed even the worlds top 'pros'.

and digital vs analog, PT vs cubendo? ill get the job done on anything, hell, ill make a mix youll be happy with with a 4 track cassette and some guitar pedals! (serious, ive done it good just like that before!) if you have an ear its amazing what you can do...

but ill have to say, ive never had a lack of processing since the athlon 2000+ in host based processing, theres no reason to make proprietary DSP now anyway, when we have quad cores or more. PTHD3? whatever. that reminds me of the claims of mac being superior to a pc, in 2001, completely ignoring the FACT that i could get a 4 cpu tyan board for the same DOLLAR amount as a 1 processor mac, and BLOW IT AWAY. power per dollar was ignored...
i can comp a 30 take vocal track while the vocalist is out for lunch in nuendo, its FAST!