# Big Question: How do you route your final summed signal?



## Waywyn (May 23, 2006)

Hello all,

i find this question pretty important and i just wanted to know, how do you guys route your final and summed signal into one stereo signal?

we all care about excellent plugins, best EQ's, the greatest sounding libraries, sensible and smooth reverbs aaaaand so on, .... but who of you ever concerned about the summing signal, the final stereo file which leaves your sequencer?

is anyone of you using external analog bus summing hardware like a tubetec summing amp or others?

i would really like to know how much the audio signals get powered and boosted up when using such things. well, those summing amps are pretty expensive but it is obvious and without a question that they bring more vitality and energy to a final stereo signal than just routing all tracks within cubase, logic or protools internally.

everyone knows there are analog tube consoles out there which cost a million of dollars, but i am 100% sure that they don't have those huge consoles to just route all signal from their mics, tracks and plugins together only because to have some real faders in their hands.

if a little project studio would upgrade to get at least some analog routing stuff involved it would be a couple of thousand bucks and i know that i still have other things to concern about  ... but sooner or later i think this would bring a real huge "uhuuu"-experience if we all had the chance to compare our stuff routed through analog stuff.

nobody can tell me that e.g. cubase is keeping the same signal flow and power routed and summed through some piece of digital code than a 3000 bucks summing amp like spl, tubetec and name 'em.

gimme your opinions, is anyone of you using summing amps and other hardware stuff for routing audio?


----------



## Nick Batzdorf (May 23, 2006)

My violently strong opinion: .

However, I have used my Millennia Media channel strips (generally the tube settings, for color) for mastering and also for adding color to synths.


----------



## Peter Emanuel Roos (May 23, 2006)

I once read comparisons of ProTools and sequencers (all digital and all often used) and there were no significant or even audible differences. 

If I were you I should look for plugins like tube simulators etc. to add some nice distortion (the correct harmonics that tubes give).

I am absolutely inexperienced in this area, so this remark may be worth nothing...


----------



## Waywyn (May 23, 2006)

Hey and thanks so far for the comments, but the thing is, that i think it is not about coloring the tracks, processing them and making them fat.

Generally i mean the whole mix process WITHIN the sequencer.

Imagine you have 16 stereo audio tracks running through a little digital sum channel of a sequencer and then ...

the same 16 stereo tracks running from the sequencer out of 16 soundcard outputs into a million dollar neve or ssl console's 16 separate inputs - just running through it, no processing of the expensive oxford eq's or whatever. just the process of summing these signals. then route this stereo signal back into your soundcard or render it to a stereo sum.

i think there might be a significant difference!


----------



## Waywyn (May 26, 2006)

Hmm seems like a lot of people just don't care about this.

We care so much about plugins, libraries and excellent gear and in the end we just put everything together through a little piece of code. Sure it might sound cool, but do we know how big, huge or wide it would sound by running through a separate summing amp, which is just build for melting tracks together?

Why there are so many sum channels to buy (and not really cheap one). If this wouldnt be an issue then all those sumchannel would be absolut unnecessary?

What do other think about that? Is absolutely in this forum nobody using such a thing?


----------



## JacquesMathias (May 26, 2006)

Hi.

Look, i've working many years at studios, but my main work is play , arrange and compose. Because of the nature of my last 10 years, i really HAD to learn mix, master. It was great to me. Actualy it take my focus to the audio thing. and...

I've recorded and produced a CD 4 years ago (i posted here like my "Brazilian Demo Reel") and i had the oportunity to record with top musicians (at Brasil) , engineers and studio. So i could see and listen the "difference" between recording in perfect acoustic rooms with expensive mics/pre-amps/cables/2" tape machine and all the hardware you can imagine and Pro Tools/Computer thing. Yes, you can listen it. There is a diference. Many people like, many dislike. 
If you ask me if your sound will be better if you route 24 audio tracks to a Neve console to mix and process everything else "out" of computer i will say : Yes, it's better. I'm not saying from any technical point of view, even because i'm a "practice audio guy", i've been training in music, not in audio. BUT, IT SOUNDS BETTER TO MY EARS. BUT, I THINK IT'S REALLY REALLY IMPORTANT IF YOU ARE GOING TO RECORD EVERYTHING IN YOUR PROJECT. I MEAN IF YOU WILL RECORD A DRUMMER, AN ACOUSTIC GUITAR, VOICE ETC...AND YOU WILL PROCESS IT TO DELIVER THE FINAL PROJECT. It will sound more "professional" to your ears in a abstract level. It does not mean that digital recording will sound bad. :wink: 


I told many things that has not many to do with this topic. Backing to the topic : I think that processing out of computer sounds better, when "what" you have out is really better than "what" you have inside of computer. If you've recorded with TOP A/D conversors, pre amps,mics and cables and you can use TOP D/A conversors to deliver to a TOP board with equalizers and individual rack compressors...dude. Even if you were a deaf person you will be able to listen the diference!  

Cheers!


----------



## Frederick Russ (May 26, 2006)

You may want to try a variety of plug-ins to compare against an analog sum-bussed mix (much like the shoot out we do here occasionally). There is a lot out there now that can add analog, tape and tube sensibilities to a mix. I had a talk with Hutch at Manley Labs about the same thing years ago. You might be able to try a high end dual channel mic pre and get similar results - but - you'll also need to route your digital audio to a great D/A converter. Are you then going to need to transfer it back into the digital domain? Maybe an Apogee Rosetta 200 2-channel 192kHz 24-bit AD/DA Converter routed to a Vintech 1272 2-channel Mic Pre (almost identical sound footprint to the Neve 1272)? API also has some sum buss options but realize that the overall effect may or may not be as noticeable as the dollars you'll be spending to make this work.


----------



## Waywyn (May 26, 2006)

Hey and thank you both for your opinions.

Frederick, i also thought about that the result might not be so huge like the dollars i would spend, but i just thought somebody is workin with summing amp stuff.

i just thought about this issue and i am always concerned about getting more loudness, smoother sound etc. (of course i know that most of this stuff is in the fingers and experience, but i am just talking technicalwise)

i heard from a few guys that there is a company named "dangerous". unfortuntely the language is just german, but you could also take the new summing amp of spl: http://www.spl-usa.com/mixdream/2384_inshort_E.html

or the tubetec thingy: http://www.tube-tech.com/?show=product& ... oductId=17

but in the end i absolutely know nobody who is using summing amps to not loose signal inside the summing bus of cubase, logic etc.

so i thought about asking around. usually guys go like: "ah, save ya money" or "dude, i couldn't live without it" (as i would say about the Inflator


----------



## Frederick Russ (May 26, 2006)

Well there is something there since some folks like to route their midi mockups through a Neve board (essentially to use its sum bus output) - like I said you might try renting a sum bus module or high end dual channel mic pre to try some experiments and let your ears decide.


----------



## Waywyn (May 26, 2006)

the only thing which really bugs me is, why every developer is doing firewire instead ob USB 2.0

since nearly all developers do firewire but every computer just got (mostly) 1 or 2 firewire ports.

uhm, did i miss something? the last thing i heard is usb 2.0 is much faster than firewire and probably ever motherboard got at least 6 usb ports.

if i am really seriously consider duende i have to make sure to buy a new soundcard as pci or so. i am still using the motu 828


----------



## Scott Cairns (May 26, 2006)

Hi Alex, there was a big deal about this on another board a few years ago. They created something called the "DAW Summing Project" - or something like that.

I think it was more specifically comparing how one daw sums compared to another, but they might've gotten into the whole analog summing thing as well.

I came very close to buying an analog summing box a couple of years back. The thing I realised though, is that you'd need state of the art A/D and D/A converters to route to your summing box or otherwise - whats the point?

Aside from that, I came to think about the deliverable format for my clients, since its for games and always in a compressed format, I didnt want to go the whole hog of analog summing for what might mightve amoutned to little or no increased fidelity.

If I was recording lots of acoustic instruments to 2 inch tape and producing a high quality CD Id consider a summing box. But I dont think its worth it for what I do.

On another (but actually related) question, what sound card do you use? Do you clock to a converter at all?

A high quality sound card with low jitter, or a card slaved digitally to a great converter, will clean up your mix, focus the bass and stereo image.

I go on about this alot, but Ill never forget changing from a mid level sound card to a RME Hammerfall clocked to an Apogee Converter. THe difference in quality was amazing. My wife even walked in the room once and commented that my music _sounds_ heaps better. (not talking about compositional aspect either.)


----------



## Frederick Russ (May 26, 2006)

Hey Alex,

What kind of soundcard/breakout box do you have for your rig? If it has wordclock I/O you may try a master clock to remove jitter - Apogee makes a cool one. Actually that may be the single one point of difference even before summing a stereo signal out to a sum bus/dual pre.

I've seen a lot of sum bus debates on the net on a lot of different boards over the years. I think one of the main things in the past that those conversations yielded was that high-end analog circuitry essentially had a "wider bandwidth" than what was offered by the digital converters at the time - but times have changed. By clocking your entire digital system with a rock solid master clock with 24-bit/192kHz resolution (like the *BIG BEN 192k Master Digital Clock*) you may be able to get a good chunk of the benefit _even before_ you get your signal to an analog sum bus. If you still want a reasonable summing buss solution, check out the *API Rack Mounted LEGACY Console* or the *MANLEY 16 into 2 Tube Mixer* for ideas. Hope this helps!


----------



## Waywyn (May 27, 2006)

Thanks a lot, 

i really appreciate since my knowledge about all those hardware things is pretty small 

the only thing i have is the motu 828 mk1 which is a firewire unit and it got only 8 in/outs but not adat connection or so.

so how would i connect a thing like the big ben to my soundcard or pc? do i have to have a soundcardw with adat connection or so?

the other thing which is pretty interesting, if the audio quality will be reduced by creating more group tracks. let say if you have e.g. (extreme example) 200 vsti tracks going into 50 groups, so we would have 4 audio tracks in each group, then you have another 5 groups which means, 10 groups (of the 50) go into each and then route these 10 groups into 1 master groups.


----------



## Frederick Russ (May 27, 2006)

Waywyn @ Sat May 27 said:


> so how would i connect a thing like the big ben to my soundcard or pc? do i have to have a soundcardw with adat connection or so?



Hey Alex,

You can connect the Big Ben Master Clock wordclock out into the 828's wordclock in (see pic).

I think its cool that you're shooting for better audio quality. I've been told by a lot of pro audio guys that using a master clock can be the single best thing to improve it where you can substantially hear the difference. I really should rent one myself to see before I buy (I have a Motu 2408 Mk II).


----------



## sbkp (May 27, 2006)

Frederick Russ @ Sat May 27 said:


> Waywyn @ Sat May 27 said:
> 
> 
> > so how would i connect a thing like the big ben to my soundcard or pc? do i have to have a soundcardw with adat connection or so?
> ...



I know lots of us play live instruments at times, where a solid clock can make a huge difference (several acquaintances of mine have had that revelations -- I haven't taken the plunge yet). But I wonder if it makes any difference for exclusively VI work. It would make what we hear on our systems sound better, but since all the mix is happening in the box, I wonder if the final WAV/AIFF output would be any different with a different clock. I kind of doubt it -- anyone here with an external clock want to do with and without versions of a mix of already-recorded tracks or VI tracks?


----------



## Frederick Russ (May 27, 2006)

We can slave our sequencer to external word clock as well as everything connected to it via adat spdif etc - this can impact everything from rendering audio to monitoring our mix. The wave and/or aiff files don't change - but the theory is that if a system is not synchronized and calibrated using a solid word clock, what we're hearing along with the aiff and wave files is the extra added jitter which can increase listener fatigue and imaging problems. It's been said that removing that jitter makes a huge difference both in the way the mixes are being monitored as well as final rendering for mastering. 

I've talked to two people who had upgraded their system with Big Ben and they were both amazed at the difference. Both said, "I can't believe this is really my gear I'm hearing!" But - we won't really know if it works for us until we either rent one to try it out on our own system or take a leap of faith and buy one.

Testimonials:

http://www.apogeedigital.com/users/index.php?show=bigben (http://www.apogeedigital.com/users/inde ... how=bigben)


----------



## Synesthesia (May 27, 2006)

I have just thought I need to reconnect my rosendahl soon - I disconnected it when I upgraded to protools HD a few years ago.. Never got round to recabling.. shame!

check out some interesting articles:

http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=28/

and a shoot out (only the mytek and digi stuff I remember, might be more now)

http://www.mytekdigital.com/compare/index.htm

The rosie made a big difference to my pre HD rig which was 882/20s. I gather its also a big diff with 888s, and not so big with HD, which was why I was lazy with getting it all set up again. Its a nightmare now of course, loads of stuff to cable together with VST farm PCs and so on.

Cheers

Paul :smile:


----------



## Waywyn (May 27, 2006)

Ah thx Frederick (and all others) for the info.

So in the end i just need to connect the big ben via word clock (or optical out?) to the motu and just go on working like everyday and the quality of my work improves "a bit" 

hope i got that right, sorry for being so stupid, but i am more into plugs and programs :mrgreen:


----------



## synergy543 (May 27, 2006)

As I understand it, jitter is only an issue when you enter or leave the digital domain going through A/Ds and D/As or when you transfer digital audio (such as through S/PDIF as opposed to just data) between two digital audio devices. Within the digital world in a computer (and between various programs and plugins) you are transfering data which doesn't change as it is being transferred. 

Here is a good article on Jitter by Bob Katz of Digital Domain:
http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=28/ (http://www.digido.com/portal/pmodule_id ... age_id=28/)

Does anyone have a link with an audio example of what jitter sounds like?


----------



## Scott Cairns (May 27, 2006)

sbkp @ Sun May 28 said:


> But I wonder if it makes any difference for exclusively VI work. It would make what we hear on our systems sound better, but since all the mix is happening in the box, I wonder if the final WAV/AIFF output would be any different with a different clock. I kind of doubt it -- anyone here with an external clock want to do with and without versions of a mix of already-recorded tracks or VI tracks?



Hi Stefan, from my understanding (and my ears), it does makr a difference. Cheaper sound cards have lower quality crystals. So when you set a sample rate, it might not output 44.1000 but 44.12432 or something like that.. I remember reading about it.

When it does do that however, you get jitter, regardless of whether you running out of your daw or not. With jitter, you get digital distortion or fuzziness, and the symptoms I described earlier; poor imaging, fuzzy bottom end.

Alex - A cheaper solution to the Big Ben is an Apogee Mini Me, you can clock via sp/dif, adat etc, and if I remember reading correctly at the time, it has the low jitter specs as the Big Ben. I believe the Big Ben is used if you're clocking a lot of things.

Currently, I slave my RME HAmmerfall via sp/dif, to the Apogee Mini Me. As I said earlier, even my wife can walk in the room and hear the difference. 

I also find I can work a lot longer, as things like violins as less harsh on the ears.

Go for it! You wont regret it.


----------



## sbkp (May 27, 2006)

Scott Cairns @ Sat May 27 said:


> sbkp @ Sun May 28 said:
> 
> 
> > But I wonder if it makes any difference for exclusively VI work. It would make what we hear on our systems sound better, but since all the mix is happening in the box, I wonder if the final WAV/AIFF output would be any different with a different clock. I kind of doubt it -- anyone here with an external clock want to do with and without versions of a mix of already-recorded tracks or VI tracks?
> ...



Weird. I wonder why an internal mixdown would need the soundcard at all.

- Stefan


----------



## Scott Cairns (May 27, 2006)

Well i dont pretend to be an expert in this, but isnt the soundcard setting the sample rate for the computer? And if I remember correctly, the sample rate is set by a crystal in the card itself. Cheap to mid level cards have cheaper crystals in them as mentioned earlier.


----------



## sbkp (May 27, 2006)

What you say is certainly true to listening to a mix. But software-based mixing is just that... software. I think an example of this is FX Teleport. If I recall correctly, the slave computers don't need sound cards at all, right?


----------



## Waywyn (May 28, 2006)

as far as i understand the slaves port the audio through lan, but when doing a final render i think you need the soundcards Khz, like scott mentioned.

so i think the apogee just sorts out jitter and the loss and brings a better quality since the soundcard (through the clock) is able to produce better wave quality. am i wrong with this?


----------



## Scott Cairns (May 28, 2006)

Thats my understanding Alex.



sbkp @ Sun May 28 said:


> But software-based mixing is just that... software.



I believe that the rendering of the file is still derived by the clock in the soundcard. (Or whatever your card is clocked too.) This is why an unstable clock makes a render so fuzzy.

BTW, Im happy to be proven wrong in this, Im not trying to make empirical statements. Just what I believe to be true.


----------



## sbkp (May 28, 2006)

Yeah, it's a good question. So... Someone with a nice soundcard and a crap soundcard... Please do a test for us! I have only a medium soundcard.


----------



## Waywyn (May 28, 2006)

oh by the way, scott, 

sorry i forgot, thx for the hint on the mini me, but call me stupid, i don't get it really.

the mini me seems to me like a preamp with some limiting and processing stuff and some of the apogee dither options.

so as for a mic/guitar preamp i plan to by the spl channel one, but on the mini me page it is nothing written about clocking the signal and preventing jitter.

or do you run your final signal of the soundcard through the mini me and then back into the computer?!?


----------



## Scott Cairns (May 28, 2006)

Hey Alex, no problems, the Mini Me is essentially a two channel A/D Converter. It also happens to have great pre-amps built in plus some compression curves and limiting available.

I use mine for two purposes;

1. As an A/D Converter for any recording in my studio (I also use it for location work.)

2. As a Master clock. I connect the Mini Me via a s/pdif cable to my RME Hammerfall card. I set the Hammerfall as a *slave* to the Mini Me. 

Im sure that at the time I researched the Mini Me, the internal gear was the same as the Big Ben, same quality crystal and so on. The Big Ben is simply giving you more options to clock devices. So that's one hidden virtue of the Mini Me, the clock in it is generally more stable than any soundcard on the market. Broad statement I know, but I did hear one engineer claim that they can tell in their studio if the Big Ben is turned off or on. They have their PT rigs slaved to it.

I can tell you one funny story about changing to the Mini me, I used to own an M-Audio Delta 44 card.

I often use the Steinway patch that Eastwest sends out for free to EWQLSO users.

When I had the Delta card, I always believed that the piano was a mono patch. THe first time I upgraded to my Hammerfall card and the Mini Me as a Master Clock, I learnt that the piano patch in fact panned from left to right;

the bass notes were in the left speaker and moved to the centre as you approached middle C. Going to the high notes, they then pan over to the right speaker.

I NEVER heard this when I was running the Delta 44 card.

If you have any other questions mate, give me a yell.


----------



## Waywyn (May 28, 2006)

*lol* funny indeed.

i had this experience back then when i upgraded my system from a soundblaster to the motu 2408 mk2 

okay, if you can use the minime as a simple clocking source, then this might help. probably i go out and rent one for a day or just try it (our big music store mostly hands out stuff which you buy and if you don't like it, just bring it back). so i can compare best 

thanks mate!!


----------



## Nick Batzdorf (May 28, 2006)

Lots of points here.

Number one: Bob Katz is a major guy who really knows what he's talking about. But his opinions are just that. For example, he says that it's much easier for a digital audio device to clock to its own crystal than to lock to external clock, therefore if it sounds better clocked externally you should complain to the manufacturer; external word clock generators don't have a sound, and you shouldn't need them in every application. He predicts that external work clock generators won't sound better than the internal clock in almost all cases, and that makes excellent theoretical sense - it takes much more work to use double PLLs (phase-locked loops) to get rid of incoming jitter, etc.

Yet someone else in this thread reports the same thing I've said many times: previous-generation Pro Tools TDM interfaces sound noticeably better clocked externally. Likewise, a friend of mine's untrained wife hears a night and day difference when his MOTU hardware is clocked by an Apogee Big Ben.

So there you go. The subtext is that every fricking discussion about digital audio in the past ten years points to Bob Katz' website! I've said many times that I've been studying this stuff for years and I still don't really understand it. Someone always points me to Bob Katz' website.  Everyone interested in this stuff should indeed read his website, but they shouldn't assume that they're experts on digital audio just because they did so.

Second point: the theory is that jitter only affects converters. Well, Thonex was at the same nerdfest I was at a few years ago where we proved fairly conclusively that that's not necessarily true. Five people independently reported the same differences between transfers made in the same set-up through different cables. Twice - we repeated the test in another studio.

So even though the difference is tiny, it's real - even though the files cancelled audibly. It makes no sense, but something is going on.

Does that mean you should get a word clock generator if you're using samples inside one machine? Well, you'll hear a difference while you're monitoring - if you have a good monitoring set-up.

Next: it turns out that machines can drift when you're sending audio over ethernet. FX-Teleport must use the ethernet port to clock to, but when I was using AUNetsend/Receive between two G5s, the machines drifted over time until I clocked them together (using the TOSlink port). The latency is still unusable, by the way, so it's not a solution.

Finally, Waywyn (was it you?): it's not that I personally don't care about the diff between internal and external summing, it's that this has been discussed and tested and discussed and tested over and over for the past five years.


----------



## sbkp (May 28, 2006)

I thought of a different way to test this: Take a wave file, load it into your DAW, set all faders to 0db, do a mixdown. Then take the mixdown, put it on another track, invert the polarity of it, and play back.

My hypothesis is that if the soundcard matters (or rather, if the soundcard has any negative impact), then there will be some signal left over, somewhere. What you _should_ get is complete silence.

In my case (EMU 04/04), I took a mix of mine and did the above. After playback, the peak meter read minus infinity. So my $100 soundcard seems to have no ill effect on an in-the-box mixdown.

- Stefan


----------



## José Herring (May 28, 2006)

Though I have no idea whether what you did is a legit test or not I will say that it seems to make sense. As long as it's all internal you're not processing actuall sound you're just transfering binary bits through different areas of running code. I can't imagine that clocking would make any bit of difference in that senario. Of course on a better soundcard the output through your speakers would be better because you're actually converting the signal at which moment timing annomalies (jitter) in conversion is a factor for sure.

But I know so many composers who have shyte for soundcards and they do just fine. It just has to be acurrate enough to mix with.

I say save the money and get a $500 card and spend the money on preamps and DA/AD converters for recording live instruments.

If you're running external mixers and more than one daw simultaneously and you need to sync lots of different external machines then slaving to black box or wordclock might be necessary. But for many of us using a main computer to handle all our audio and using FXTeleport for additional machines or just using machines as outboard sample boxes with midi over lan and lightpipe, slaving to worldclock seems like overkill just for audio purposes.

Jose


----------



## Nick Batzdorf (May 29, 2006)

sbkp, that's what we call the phase inversion test in the art world. While X + -X = 0, just because you get complete phase cancellation doesn't mean a shitty sound card is perfect.


----------



## Synesthesia (May 29, 2006)

Nick Batzdorf @ Sun May 28 said:


> Lots of points here.
> Does that mean you should get a word clock generator if you're using samples inside one machine? Well, you'll hear a difference while you're monitoring - if you have a good monitoring set-up.
> 
> .



Very interesting post Nick - thank you for your comments, I have also been studying Bob's opinions on Jitter!

Interestingly, his theory would still hold up on your remark above - if I understand you correctly - in that your final DA is the one that will react to any jitter in the chain, and ..provided.. that no jitter affects the transfer of digital audio from one Digital device to another, the only effect of the Jitter is in your listening to the playback.

So, one might conclude that in a system where you are using all digital transfers of audio from slaves to master DAWs - but reliable digital transfers, ie not Network audio, as that introduces a time effect - but Optical, AES etc, the digital audio 'packets' are all transfered, and the jitter is eliminated naturally at each transfer and storage of the packet.

Thus, if you are mixing in the box and delivering a digital file of the mix by (for example) FTP, jitter will have absolutely no effect, and thus you needent worry about it ---- **even if you can hear it**.

It is only where you have AD or DA where the jitter may degrade the audio. For the final DA, if its just for listening and not to master from, who cares. For any AD, you just need to make sure you have a decent crystal, or wordclock (a second best to a decent crystal).

So, also, the audible difference for people with all in-the-box systems, ie no console etc, is just that the playback is less jittery. Not a great reason for someone on a tight budget to lay out a grand on a wordclock generator!!

Cheers

Paul


----------



## kid-surf (May 29, 2006)

Is this a wordclock thread? :mrgreen: 

Well, I use a master clock, the Genx96. Keeps everything on the same clock. Has enough I/O for mr rig.

THis stuff is really subjective. So keep the receipt I say. If you're hearing an improvement keep it, if not don't. But I am surprised you guys with multiple clocks aren't getting clicks and pops, or maybe I'm misunderstanding they way some of you are set up.

BTW --- it may have already been said, but: Unless you have many things you need to clock, you shouldn't assume that a clock will make an improvement. 

As always "use your ears".........

cheers


----------



## Scott Cairns (May 29, 2006)

I find the low jitter I get by locking to my Apogee an added bonus, I noticed that i can work far longer before my ears get fatigued. So in a way, my Apogee pays for itself on that alone. 

But I didnt get it just for clocking, I use mine for the A/D conversion of anything I record and THAT makes a huge difference. I sometimes record actors from home and have had clients comment on the quality of the sessions.


----------



## sbkp (May 29, 2006)

Nick Batzdorf @ Sun May 28 said:


> sbkp, that's what we call the phase inversion test in the art world. While X + -X = 0, just because you get complete phase cancellation doesn't mean a shitty sound card is perfect.



Nick,

I agree totally. I'm not saying a shitty card (just what are you insinuating, sir!?!?!?  ) is perfect. I'm saying that I don't believe the card's clock is used for doing in-the-box mixdowns. It wouldn't make sense to do so.

Another way to test... Mix a project twice. The files should be identical. If they're not, then something is amiss, and it could be that the soundcard is being used (could be lots of other things, too). This test has far more variables, though. It's possible, for example, that some effects algorithm has some randomness in it. Or in my case (haven't tracked this one down yet), the SIPS Legato script produces different results on different plays.

How it sounds coming out of the speakers is a different ballgame altogether.

- Stefan


----------



## Nick Batzdorf (May 29, 2006)

Just to be clear, I didn't mean I've been studying Bob's opinions on jitter for years, I mean I've been nerding about digital audio for years - A/B-ing stuff, etc. In all honesty, I lost interest a while ago, but my point is that there are a lot of strange phenomena with digital audio that not a lot is known about.

Paul, the theory is that jitter only affects converters - D/A or A/D - so transfers in real-time or not real-time either succeed or fail. By that logic the clock shouldn't matter for anything other than the A/D (if the music is going to be played on lots of other systems), and things like cable make absolutely no diff.

I don't believe that, as I said. Also, I don't believe that audible phase cancellation tells you everything. One of the people at the nerfdfest above is George Duke's engineer, and he said, "Oh, the phase-cancellation test? That's what the CD reproduction houses always use to try and convince us that their CDs are identical to what we sent them, when we can all hear perfectly well that they don't sound the same."


----------



## Nick Batzdorf (May 29, 2006)

Real-time audio is clocked to whatever you tell the system to use, but my layman's guess is that offline bounces (faster-than-real-time bounces) use the computer's clock.

Why you can sometimes get glitches with off-line bounces is another question, and I don't know the answer.

Also: clocking the whole system from the same clock is one issue. You absolutely must do that or you will get lots of clicks and pops with 100% certainty. Every time you transfer digital audio from one machine to another - and that includes digital effects units, recorders, computers, or anything - they have to reference the same clock.

We're assuming the clicks and pops are gone; the way you distribute the clock is the issue I've been talking about. You certainly don't need any extra pieces of equipment to get rid of clicks and pops.


----------



## sbkp (May 29, 2006)

Nick Batzdorf @ Mon May 29 said:


> I don't believe that audible phase cancellation tells you everything



Nor do I. Just for the record, my test was *not* an audible phase cancellation test. As I said in my post, the resulting level was minus infinity the whole way through. That means that not a single bit was different.

- Stefan


----------



## Nick Batzdorf (May 29, 2006)

I think that's what it means. But audible cancellation should tell you the same thing, in other words you shouldn't be able to hear the odd bit that's different - if two files cancel audibly, they should be virtually identical.

Except they aren't, and I don't have an explanation for that.


----------



## sbkp (May 29, 2006)

I have a feeling we might be talking about two different things. 

All I'm saying is that I don't believe the sound card's clock is used during an in-the-box mix. My cancellation test doesn't only product no audible results -- I'm not talking about virtually identical files. I'm talking about bit-for-bit, exactly the same files. There is not a single bit out of place after my test. I'm not testing that only with Cubase either. I wrote a program to scan through the file for non-zero bytes, and there are _none_.

So my conclusion is that either my $100 sound card has an incredibly stable, totally predictable, zero-jitter clock (yeah, right), or that the sound card's clock isn't being used for mixdowns.

- Stefan

P.S. It's not technically correct to call it a phase-cancellation process. The polarity of one signal is reversed, and that's _not_ the same thing. But yeah, I know... Everyone uses those two terms interchangeably.


----------



## Nick Batzdorf (May 29, 2006)

Yes, yes, phase cancellation and polarity reversal are two different things. There's also a difference between can and may, and you're supposed to put commas everfrickingwhere. 

As I said before, to me it only makes sense that off-line/unreal-time bounces use the computer's clock. The sound card's clock - or the clock it's clocking to - can only be used for real-time audio.


----------



## sbkp (May 29, 2006)

lol...  


Okay, good. We agree. Everyone just move along now... Nothing to see here.


----------



## Mike Greene (May 30, 2006)

Nick Batzdorf @ Mon May 29 said:


> . . . George Duke's engineer, and he said, "Oh, the phase-cancellation test? That's what the CD reproduction houses always use to try and convince us that their CDs are identical to what we sent them, when we can all hear perfectly well that they don't sound the same."


That quote right there sums up why I'm so skeptical of listening tests where a bunch of heavyweights with golden ears all agree on some conclusiuon that I don't think makes theoretical sense.

If the replication house honestly performed a test that proves (via an x + (-x) test) that the CD is bit for bit the same as what they were sent, then how is it that "we can all hear perfectly well that they don't sound the same?" CDs don't have word clocks and CD players have buffers, so unless the CD gave errors, a file is a file. If the 1s and 0s are all in the same place, they should sound the same.

Maybe ProTools or Peak screwed up the file import from the CD. (Hopefully Duke's engineer didn't just compare the output from a CD player with the output from ProTools playing his source.) Maybe the replication house's test wasn't honest so maybe Duke's engineer should have verified it himself.

Years ago, Bernie Grundman said in a magazine that he didn't use digital transfers because they always sounded different. In other words, he could hear a difference in 1 to 1 digital clones. I couldn't, but I guess he could. Or more likely, believed he could.

Once at an AES show, a medium name guy tells me to check out the new Lawson mic. "It sounds exactly like a U47! The only difference is it sounds better!" Consider how ridiculous that statement is. And the guy's no moron!

One time I spent about 10 minutes perfecting the EQ on a vocal track. I nailed it. Only to later notice the EQ wasn't switched in. I was twiddling inactive knobs. Meaning I don't even trust my own opinions anymore unless they're scientifically done.

I've heard such rave reviews of the Big Ben (master word clock source) that one day I'll buy or borrow one and check it out for myself to see if it improves my Digi 192. But even that test will be difficult because since word clocks can't quickly reset, A/B-ing will be slow and therefore more subject to other variables.

I'll even invite Nick and his friend's wife. And might as well buy some Monster cable to put that debate out of it's misery too.

- Mike Greene


----------



## José Herring (May 30, 2006)

Mike Greene @ Tue May 30 said:


> [
> One time I spent about 10 minutes perfecting the EQ on a vocal track. I nailed it. Only to later notice the EQ wasn't switched in. I was twiddling inactive knobs. Meaning I don't even trust my own opinions anymore unless they're scientifically done.
> 
> - Mike Greene



Ha! Yeah we've all been there. There's a certain amount of voodoo mysticism in what we do and I've found that getting rid of the "everybody knows" or "leading expert agree" data in my head, has led to better music.

I once had a guy in a music store telling me that I should buy only apogy CD's and burn at 1x speed to get a deep groove of binary bits in the master CD. :roll:

I remember once I was doing a clarinet take and I was driving the engineer crazy trying to get "the perfect take". We did about 10 or 12 takes of my solo. Just to prove me wrong he put up all ten takes simultanously and said listen. So I listen and to my amazement each take played together perfectly with out any flanging or staggard attackts. I had laid to tape 10 times in a row the exact same solo.

From that moment on I realized that there's the real world and then there's the feable voice in your head telling you that it all sucks, when in the real world there's no difference.

I suspect the same thing is happening in this debate.


----------



## Nick Batzdorf (May 30, 2006)

Mike, there are two sides to this. One side is yes, you can fool yourself. The other side is that nobody is going to tell me that what I hear perfectly well isn't real.

Both sides are valid at different times. In this case, we all heard the same thing independently without any possibility of anyone cheating. There's no question that there were differences in the transfers; this wasn't a subjective listening test of something that we like and you don't or v.v.

However, it was a very, very subtle difference - well beneath the point where anyone should give a flying hoot. So nobody came to the conclusion that we cared about the difference, just that there shouldn't be any and there was one - and it made no sense. Of course 1 - 1 = 0. But that doesn't mean the bits were playing back the same, in fact I can tell you they weren't. You're free to believe me or not; it makes no difference to the reality. There's obviously an explanation, of course - I just don't know what it is.

Thonex was there, in fact the first test was at his studio. He'll tell you I'm not nuts. Well, he'll tell you I am, but not about this.

By the way, the Big Ben was driving my friend's MOTU boxes, not a 192. The 192 is a higher-end box, so it may not make the same difference. My friend is beyond obsessive - he's an excellent engineer and I trust his ears.


----------



## Mike Greene (May 30, 2006)

Nick Batzdorf @ Tue May 30 said:


> nobody is going to tell me that what I hear perfectly well isn't real.


Oh yeah? Well, what you hear perfectly well isn't real! So there!!!

:mrgreen: 

You may very well be right and you're no dummy, so I won't make any final conclusions yet. But given all the various bogus hocus pocus things I've heard over the years, until I do my own tests, or someone shows me the math, color me skeptical.

I have a couple MOTU Travelers which should be similar to your friend's converter (although it's his wife I'm more interested in.) I'll test that along with the 192 versus the Big Ben. You'll definitely be invited if I ever do it.  

- Mike Greene


----------



## Nick Batzdorf (May 30, 2006)

Sure I'm a dummy! And you're right to be skeptical.

Thanks for the invitation. I'll ask my friend's wife if she wants to fly down from a small resort in Oregon.


----------



## Mike Greene (May 30, 2006)

Without her husband, of course. Hmmm . . . in which case I may have to retract your invitation as well.

- Mike Greene


----------



## Nick Batzdorf (May 30, 2006)

But what if she's a total bowzer?


----------

