while I am not an AES genius or anything - I have to say that on just about every platform I have worked on, ProTools, Logic, DP or Nuendo - mixing out multiple busses instead of just the Two Buss - always sounds better. If I understand it correctly there are several things happening that are all realated. The first is that by going out multiple busses you can send more level of drums to one buss , guitars to another, Keys, vocals, loops- - what have you. Thus you are using more of the information from the original track. I have been led to understand that to really get the benefits of 24 Bit you need to keep your levels up between 6 and 0 digital max. SO - first off now you have more resolution being sent out more places - Plus on a single Two Buss - that is alot of number crunching - your computer is already doing disk ops, screen re-draws, summing, running plugs, blah blah blah - SOMEWHERE it has to start lopping off numbers (32Bit float helps alot)- this is where TDM really fails. Digital mixers do like 4 things(ok more) - easy to keep alot of horsepower where it is needed - number crunching. Yes the algorithm for summing is very important - and I know very little about the different ones - again where TDM fails - math made for computers in the early '90s - anyone else here remember when you had to guess at your EQ - then process the track before you heard it? I have actually been sending all my mixes out separate Busses or Auxes before they go to the Two Buss - or to a console. It works great for the UAD card and the Firium EQ as I can add sample delay to other tracks to keep my stuff lined up - plus I keep the buss faders down a few db that way I can push up the original track and get as much resolution as possible. Not real technical - but it works for me - and I believe it does make a difference. Oh yeah - and keeping everything at 44.1K 24Bit. No sample rate conversions please. aaron

fixnmix2k wrote: > The first is that by going out multiple busses you can send more > level of drums to one buss , guitars to another, Keys, vocals, loops- - what > have you. Thus you are using more of the information from the > original track. I have been led to understand that to really get the benefits of 24 Bit you > need to keep your levels up between 6 and 0 digital max. SO - first off now > you have more resolution being sent out more places - Absolutely negligible. And then you have some full scale stems which have to be turned down when summed somewhere else. What have you gained? > Plus on a single Two Buss - that is alot of number crunching - your > computer is already doing disk ops, screen re-draws, summing, running plugs, > blah blah blah - SOMEWHERE it has to start lopping off numbers (32Bit > float helps alot)- this is where TDM really fails. Bullshit. TDM is 48 point fixed now. No problem at all. And how do you think the summing algo works when the computer strains? "Ah, well, I think I let some numbers go because I should give some power to the screen redraws..." If there´s not enough DSP power you get an error message, but no compromised summing. > the algorithm for summing is very important - and I know very little > about the different ones - again where TDM fails - math made for > computers in the early '90s Ah, yes, math underwent some major changes since the early 90s... > I have actually been sending all my mixes out separate Busses or Auxes > before they go to the Two Buss - or to a console. It works great for > the UAD card and the Firium EQ as I can add sample delay to other > tracks to keep my stuff lined up - plus I keep the buss faders down a few db that way I > can push up the original track and get as much resolution as possible. Yes, lose some here, gain some there... You should practice proper gain staging on ALL platforms. Peter --- http://www.merlinsound.de

Hi Aaron, Thanks. To sum up your argument: my O1V can dedicate more resources to summing and therefore is better, and it potentially has a better algorithm. This is what my ears tell me, but I still find it perplexing. Seems like the math would be out there and Logic would have it. Maybe it's a resources thing, but I feel that this would still happen in Logic with only 4 stereo tracks which would barley take any processor load. The Yamaha has way more headroom too. When I send all the tracks separately out to the mixer, with all inputs at unity, I have headroom to spare and I usually turn it up. In logic I would need to turn it down 5-6 dB. Hmmmm. Thanks, - Ben

kingumfufu wrote: > The Yamaha has way more headroom too. When I send all the tracks > separately out to the mixer, with all inputs at unity, I have > headroom to spare and I usually turn it up. In logic I would > need to turn it down 5-6 dB. What do you call "unity" here? Are you sending the signals digitally to the Yamaha or via analog I/O? Peter --- http://www.merlinsound.de

<Admin - please let us not get bogged down in another thread about different sounds between different digital systems. This has been discussed here and on many other forums/lists ad nauseam. Thanks> Have I offended Peter? - sorry if I had - the tone of the reply was a little alarming - I am not trying to mislead anyone or kick some other platform inthe teeth. I work with ALL the DAW's it is my professional job. I can hear the differences in them - and it does come down to a lot of things - great original source, good performance, lots of bits used up, and good math in a DAW. ;) just to name a few. --- In logic-users@yahoogroups.com, "Peter Duemmler" <merlin@m...> wrote: > > SO - first off now > > you have more resolution being sent out more places - > > Absolutely negligible. > And then you have some full scale stems which have to be turned down when > summed somewhere else. > What have you gained? What I have gained is as much of the original digital information as I can have. Then I am not just turning up 'empty bit space' further down the line. I started working on analogue tape when I first began my career - so I understand that if I get enough level on tape - I don't hear as much tape hiss - in digital - if I don't get enough level it does not sound as good (brittle top end - stereo image suffers) - GRANTED there are a lot of other factorsbut the principle is similar - heck CD players use error correction to 'guess'at what belongs in certain places when information is lacking(ok not totally a guess but it is merely for a point). Thus: I am getting as much information to everywhere that I can form the orginal source. > > blah blah blah - SOMEWHERE it has to start lopping off numbers (32Bit > > float helps alot)- this is where TDM really fails. > > Bullshit. > TDM is 48 point fixed now. No problem at all. > And how do you think the summing algo works when the computer strains? > "Ah, well, I think I let some numbers go because I should give some power > to the screen redraws..." > If there´s not enough DSP power you get an error message, but no compromised > summing. I have failed to explain my thot on that thoroughly. That was my fault - Iknew what I was going for but failed to explain it properly. I know that if you computer re-draws the screen it does not stop processing audio - or just magically forget a bunch of information to make time and CPU cycles for it - the point is that every program is made to be efficient so all tasks can be done - and it is all, of course, math. > > the algorithm for summing is very important - and I know very little > > about the different ones - again where TDM fails - math made for > > computers in the early '90s > > Ah, yes, math underwent some major changes since the early 90s... I think there are a lot of people on this list that will agree that the the "Pitch-n-Time" in Logic does not sound nearly as good as a lot of other programs. They have all adopted newer and different algorithms - which use math - oddly enough. The same with how your audio program crunches down all that information to a stereo mix. That is all I was saying. I know a couple of guys whom re-wrote the Digimixer for PT TDM. Their mixer engine sounded so much better than the original Digi mixer is was frightening. (www.digitalaudiomiracles.com) They were suggested by Digi to stop - of course - since PT is a protected product and they monkeyed with the code. But it clear proof that there are better ways. > > I have actually been sending all my mixes out separate Busses or Auxes > > before they go to the Two Buss - or to a console. It works great for > > the UAD card and the Firium EQ as I can add sample delay to other > > tracks to keep my stuff lined up - plus I keep the buss faders down a few > db that way I > > can push up the original track and get as much resolution as possible. > > Yes, lose some here, gain some there... > You should practice proper gain staging on ALL platforms. > > Peter > --- > http://www.merlinsound.de Well, I think I am talking about a good gain structure all the way thru. I don't want this to be a pissing contest. I honestly believe that it reallyall comes down to a great song - folk, pop, rock, makes no difference. <Snipped by admin> We can all sit here and nitpick each system - bit depth - latency issue - sound card - A/D D/A converter - clock - and they all do makea difference. But there is more to it - There are a lot of ideas and opinions - these are some that I hold from personal experience, some from working with people whom help decide the actual digi standards, and others from working with people who put different methods to work everyday. This is what I love about this group - I learn so much all the time. Thank you all very much. aaron

> > The first is that by going out multiple busses you can send more > > level of drums to one buss , guitars to another, Keys, vocals, loops- - > > what have you. > And then you have some full scale stems which have to be turned down when > summed somewhere else. > What have you gained? Well, if your intent is to sum analog, you'll have a higher s/n ratio in the analog domain. You probably know this, if so, it's for everyone else: routing a full scale signal into a console means you'll be turning the input trim way down on your console's input strip, which is a vastly better idea than feeding your console a soft signal (such as it might be if you were summing 'in the box' then simply routed it to dedicated outputs) and using the console's gain as makeup, which will add some amount of noise, potentially a lot. And then there's the camp who say that the more you pull back a digital fader, the more damage you do to the signal. I personally believe that's the legacy of math from a bygone era, YMMV. > You should practice proper gain staging on ALL platforms. Wasn't that what he was suggesting? u b i k