Before it gets going:
Most converters sound close enough under controlled testing. The place converters show their true benefits happens when you add the interface suited to the workflow. The driver around it is just as important, as is the software and added sections like optional ways to gain stage. :cool: Its a lot more than just the controlled test.

Enjoy.

audiokid, post: 428618, member: 1 wrote: Before it gets going:
Most converters sound close enough under controlled testing. The place converters show their true benefits happens when you add the interface suited to the workflow. The driver around it is just as important, as is the software and added sections like optional ways to gain stage. :cool: Its a lot more than just the controlled test.

Enjoy.

that's what i was thinking. so if someone has a work load of 16 to 24 tracks on the DAW and they bring 8 out to a sum mixer and then on to an uncoupled DAW /recorder, will firewire and USB3 do the trick? i suspect there are many who are doing even more than that with no problems.

Kurt Foster, post: 428619, member: 7836 wrote: if someone has a work load of 16 to 24 tracks on the DAW and they bring 8 out to a sum mixer and then on to an uncoupled DAW /recorder, will firewire and USB3 do the trick?

absolutely.

audiokid

here's what i'm thinking. my i/o26 has inserts on the 8 mic pres. i take my JLM TMP8 and use it's 8 pres into the inserts on the i/o26. that puts the pres straight into the converters.

at mix, i bring 8 out of the i/o26 into the LittleOne sum box and 2 out into 2 channels of the TMP8 or if i can score a second one, 2 ORIGIN STT-1'(s) and on to the 2 mix converters though line ins on the AG 06, so any gain made for preamping and make up is solid Class A X former balanced. this should actually sound pretty good. what do you think?

Here's where I think my train went off on a different track than the other posters on the last thread - I was referring more to a basic i/o situation, and not as much a hybrid workflow.

I was speaking of a different situation, in that my curiosity was based on a more simple situation of getting a signal into a DAW .... Point A to Point B.

If I have a nice pre, like say, the ADK, but which has no digital i/o at all, and hence no converters, and I come out of that preamp into a converter, what are the benefits or detriments of using something like Presonus's converters vs using a higher end conversion system, such as a card-based model, or a standalone rack mount unit? Or, as Kurt mentioned, if I have an analog console, and I'm recording a live band and I need to send 12 channels of audio at once through 12 converters to get into a DAW, what is the best way to go about doing that in a way to best preserve the integrity of the audio?

I'm not slyly making suggestions here, nor am I making any point... these are actual questions. I'm not hinting at anything, nor am I underhandedly trying to suggest anything.

DonnyThompson

don't take this a gospel. i'm as much in the dark as anyone. but i'll tell you what i think i have picked up. anyone please feel free to correct me.

if you need to record high track counts, 16 or more tracks live in one pass, the best performance will come from a higher end converter. as track counts increase and plug counts rise with larger sessions the benefits of better implementation come into play. pci systems, MADI and ethernet are the tickets to stability with tracking live with many tracks. recording 8 tracks at a time or less it's not such a big deal unless you're already monitoring 32 tracks with plugs. i think the one thing that is a tell is what kind of latency performance your computer is yielding. it's about data through put.

Thunderbolt promises near latency free recording at high sample rates but so far we forced into the Apple orchard for that luxury.

Kurt Foster, post: 428616, member: 7836 wrote: ok you guys said you wanted to get together to do a converter shootout. what converters do you propose to use? can you do a double blind /averaging test?

There's no need to reinvent the wheel. I already did this in my [="http://www.ethanwiner.com/converters.html"]Converter Comparison[/]="http://www.ethanwin…"]Converter Comparison[/]. The article and video explain how this is done reliably, letting you compare three sound cards / converters from a $25 SoundBlaster to a high-end Lavry model. This comparison uses five tracks, though the notion that "stacking" occurs is disproved in my [[url=http://="http://www.youtube…"]AES Audio Myths[/]="http://www.youtube…"]AES Audio Myths[/] video starting at 28:28.

--Ethan

Ethan, it's to bad you would say yes and then renege.

i would prefer to see a new test with high sample rates 96 /192 ... your tests use 10 year old converters at 24/48. those tests are outdated at best.

also a lot of us just don't want to punch into your site Ethan. we don't start these discussions here just to generate traffic for your site. this is one reason you experience such animosity here, your constant attempts to direct RO traffic to your site. if you don't wish to participate here, perhaps someone else will. i might even try this myself BUT to be honest my health issues make any work difficult. i just don't have the stamina i used to. i'm working on trying to regain a bit of strength.

What the hell? Drive traffic to my site? I spend a huge amount of time and effort to educate people, write articles, make videos, and create audio examples. I spent half a year on my two AES videos, and I never made a dime (or even tried to). Other than occasionally getting a few hundred bucks from a magazine, the vast majority of what I do earns me a grand total of $0.00. So by your logic, even though my website contains a textbook perfect project that explains exactly what is being asked here, for some reason I shouldn't link to it?

That said, any imagined benefit of sample rates higher than 44.1 KHz has been debunked soundly many times. To believe that my test is somehow invalid because it's not at 96 KHz is chasing unicorns. But I'll be glad to go to your web site and read your article and listen to your files when you do a proper comparison and post the link here.

--Ethan

Ethan Winer, post: 428636, member: 1430 wrote: What the hell? Drive traffic to my site? I spend a huge amount of time and effort to educate people, write articles, make videos, and create audio examples. I spent half a year on my two AES videos, and I never made a dime (or even tried to). Other than occasionally getting a few hundred bucks from a magazine, the vast majority of what I do earns me a grand total of $0.00. So by your [[url=http://[/URL]="http://www.apple.co…"]Logic[/]="http://www.apple.co…"]Logic[/], even though my website contains a textbook perfect project that explains exactly what is being asked here, for some reason I shouldn't link to it?

there's no reason that could not be copied and pasted here embedded in RO but you want traffic to be diverted from here to your site. why?

Ethan Winer, post: 428636, member: 1430 wrote: That said, any imagined benefit of sample rates higher than 44.1 KHz has been debunked soundly many times. To believe that my test is somehow invalid because it's not at 96 KHz is chasing unicorns. But I'll be glad to go to your web site and read your article and listen to your files when you do a proper comparison and post the link here.

no benefit in moving the filters higher? no improvement in jitter by increasing resolution? this is why Ethan. you make the stupidest assertions based on your own assumptions, observations and nothing else. 96K is chasing unicorns? i hope not because 24/48 really still sounds like ass. you know better than the whole world?

i should have known this would happen. i knew it was a case of strange bedfellows when i suggested this. on the other hand, you said you would do something and then you punked out. i hope you conduct your business in a better way. well so much for this ....... Chis you were right.

Ethan Winer, post: 428636, member: 1430 wrote: I'll be glad to go to your web site and read your article and listen to your files when you do a proper comparison and post the link here.

lol ... i don't have a web site. but then again i'm not endlessly promoting myself so i can sell overpriced products to the gullible.

Meanwhile, back to the subject of this thread...

When assessing converter boxes, it's important to keep separate in your mind the conversion qualities and the interfacing qualities. The top converters excel in both. I purposely didn't contribute to the fisticuffs in the other thread, but it's my opinion that these two things were being argued over as though they were one and the same. Note that I haven't distinguished between input (ADCs) and output (DACs), as the reasoning applies to both.

One of the points of contention was concerned with the sonic performance of the converter chips themselves. The chips do indeed limit the ultimate performance possible from a design using that chip, but there is so much else to the engineering besides the chip itself that affects the overall quality of the conversion. This list would include conditioning circuitry, clocking, power supply cleanliness, interference rejection and a host of other things that you would not believe have an effect on the result. It goes without saying that you can take a converter chip capable of the highest quality conversion and get a bad result from it by poor engineering. Unfortunately, the reverse is not achievable, that is, you can't turn an average converter chip into a star performer by throwing engineering effort and expense at it.

All this is independent of how the converter boxes are attached to a computer for input and output data transfer. In this regard, there is an absolute measure, and that is whether the data I/O happens without error (data or timing) under all conditions.

Boxes that have poor sonic performance can have exemplary I/O performance, but it is usual that the better quality sonics come from designs where the interfacing is above reproach as well. To achieve this, it is necessary that the sale price of the unit reflects not only the higher production costs but also the necessary development budget, so all-round quality is never cheap. I've probably told the story before of a design commission I had for an audio interface which resulted in a lovely-sounding box at the pre-production level. Then the manufacturer's purchasing department got hold of it and for the production units bought the lowest-cost "equivalent" amplifiers and other components in place of the ones I had carefully selected over many, many hours of listening tests. The result was to turn a nice-sounding design into a mediocre one. I refused to do work for that manufacturer again.

Kurt Foster, post: 428621, member: 7836 wrote: here's what i'm thinking. my i/o26 has inserts on the 8 mic pres.

I admit I'm a novice it term of Highend converter comparaison but my eyes went astray when I read pres in converter tests.

May I suggest, you all take the same already recorded tracks and use them either in a round trip or unit to unit transfer and then compare with ears and reverse polarity to detect differences..
Or just tell me to go away ;)

pcrecord, post: 428641, member: 46460 wrote: I admit I'm a novice it term of Highend converter comparaison but my eyes went astray when I read pres in converter tests.
May I suggest, you all take the same already recorded tracks and use them either in a round trip or unit to unit transfer and then compare with ears and reverse polarity to detect differences..
Or just tell me to go away ;)

Marco,
i was describing my signal flow to Chris, not proposing a method of testing.

i would rather hear the AtoD as well as DtoA .... imo, we need to record a full band .... drums bass guitars keys vox .....

Kurt Foster, post: 428621, member: 7836 wrote: audiokid

here's what i'm thinking. my i/o26 has inserts on the 8 mic pres. i take my JLM TMP8 and use it's 8 pres into the inserts on the i/o26. that puts the pres straight into the converters.

at mix, i bring 8 out of the i/o26 into the LittleOne sum box and 2 out into 2 channels of the TMP8 or if i can score a second one, 2 ORIGIN STT-1'(s) and on to the 2 mix converters though line ins on the AG 06, so any gain made for preamping and make up is solid Class A X former balanced. this should actually sound pretty good. what do you think?

It sounds like it might be cool but this is always very subjective , per what Bos pointed out. Thanks for chiming in Bos. I wish you would have the moment Ethan dropped the links but it is what it is. I had a lot to offer in that thread, had it not become Ethan's moment to shill that ...

Kurt, group stems and keep the whole mix all together as it leaves and returns. I capture the entire mix on a second DAW or capture device because its easier to sum like that. You also have to disable your master bus completely because you don;t want your master bus bleeding into the stem channels. Do you know what I mean.

Donny did this on his first attempt. Some guys mix OTB for years not realizing they are sending stems and the master bus out togther. It makes everything phasy.

Best thing is to try it and start listening. Post your mixes. Some need AES interfacing others can get by on USB interfaces. Its all subjective to your workflow of each session.

No matter what you do, I believe the entire mix needs to go OTB together , not just a few stems. Does that help?

Back to this topic, I don't want to be summoned here. I have no interest in the converter comparisons Ethan did. They are ridiculous. I use good converters for a variety of reasons. None of the reasons Ethan and his group address.

Ethan Winer, post: 428636, member: 1430 wrote: I'll be glad to go to your web site and read your article and listen to your files when you do a proper comparison and post the link here.

audiokid, post: 428651, member: 1 wrote: Kurt, group stems and keep the whole mix all together as it leaves and returns. I capture the entire mix on a second DAW or capture device because its easier to sum like that. You also have to disable your master bus completely because you don;t want your master bus bleeding into the stem channels. Do you know what I mean

yes i know what you mean ... i have been doing this same thing for ten years only summing through the Mackie ... and recording to a CDr ....

audiokid, post: 428651, member: 1 wrote: Best thing is to try it and start listening. Post your mixes. Some need AES interfacing others can get by on USB interfaces. Its all subjective to your workflow of each session.

the multi track converters are firewire ... the mix 2 track will be usb3 i hope ... then again, i may just stay itb if it sounds ok ...

It should be fine.

I prefer capturing to another DAW for a very obvious reason which I will be happy to share once you are all familiar and ready to start detail listening. We can have fun going through some online audio comparison which might be interesting for a lot of us.

yes i will be going to a second daw .... i said cdr because that's what i used in the past. i want to create masters at higer sample rates than 44.1

Set the capture DAW to the destination SR and you are golden.

regarding the 44.1 . Maybe Bos can chime in on this particular point.

Example: A Lavry Black at 44.1 sounds better to me than example, a FF800 at 44.1. I've noticed better converters sound better at lower SR. I've noticed poor converters sound poor at all SR. So, if you are conserving DATA and have a DAW that runs better at 44.1, a better converter is a better choice.

I track a fare amount at 44.1. Mind you, they are projects where I know they will be deleted after the show is done but they still sound great in my DAW system. But I also use Samplitude Sequoia so maybe that's part of the subjective...
All the SR I choose on good converter sound pleasing enough.
Good converters sound good at all SR to me. They just lack some of the luster and fuller, open sound we all hear as better. I don't disagree on a comparison done at 44.1. I just don't think its of any value doing any comparison for the sake of hearing which one is better in a round trip.

I did round trip test years back where I took mixes and looped them a few times like Ethan. At that point I knew it was best to never repeat the same SR twice . Round trip
In my blunt words..., a round trip is most wrong thing you could ever do to your mix, especially when you are investing thousands into gear and technology, all to loop it back to the same session like these guys do in tests and how most people capture their hybrid mix.
That insanity started with Fletcher and Mixerman years back.

The smart ME figured this out and have been making a living at capturing mixes on DSD or a DAW for years. Some come by it innocently because we give them our mixes and they pass it through a matrix on a second system, which is exactly what I do but all in one pass. ;) Two DAW's uncoupled. Think about that...

In 2006 when converters finally improved and hybrid multitracking actually started becoming realistic, that's when I started thinking about capturing a big mix like the ME. Thus, stem mixing and mixing into a master.
Once I got off Pro Tools, everything started sounding better but one thing I never do is round trip processing. To my ears... its the best way to degrade your sound. Or maybe I should say, to make a Lavry Black sound like a Behringer. Which is what Ethan is saying but not knowing how to avoid.

I'm more interested in the tracking and monitoring part. I think the most typical way to mix these days is ITB. as you add more and more process and gear to it a comparison becomes more localized in nature.

All I could offer is a comparison of live tracks from the studio, where the same instruments where all tracked thru a motu an ensemble and a Rosetta. I guess it'd have to be 8 or 16 tracks. I dunno if this is the kinda thing you were getting at Kurt?

in all my subjective tests from my crappy gear to some decent pro standard stuff, the most expensive stuff almost always sounds better. Cheap stuff rarely outperforms the expensive. There are plenty of times where cheap and dirty is much more appropriate. (I've scrapped studio recordings of the same songs, for the basement reel to reel 4 tracks) .

I've always had 'expensive taste' and most often when I'm looking at anything from shoes to technology, to whatever, I always tend to like the stuff, and then see it's the priciest. Lol needless to say, I do t have much.

What I'm getting at really, is what are you questioning Kurt? Whether it's better, or by how much, or how much a is than b relatively.? Not being a jerk, asking a calm, non aggressive question lol

kmetal, post: 428664, member: 37533 wrote: All I could offer is a comparison of live tracks from the studio, where the same instruments where all tracked thru a motu an ensemble and a Rosetta. I guess it'd have to be 8 or 16 tracks. I dunno if this is the kinda thing you were getting at Kurt?

that would be killer Brian. can you do at least 96 or better 192? that's part of what i was wondering ..... if higher sample rates narrow the differences ...
also wouldn't it would be best if they were tracked simultaneously, on the same type of daw /computer combination?

kmetal, post: 428664, member: 37533 wrote: What I'm getting at really, is what are you questioning Kurt? Whether it's better, or by how much, or how much a is than b relatively.? Not being a jerk, asking a calm, non aggressive question lol

i want to see how much difference there is between say the Rosetta and the MOTU ...

thank you.

It's Kyle. Lol.

Sure I'll see what the track/count limitations are for the gear. Higher smal,e rates are my current area of ineterst as well. I'll do as much live as I can. And also, I'd like to try two different daws. We have pt, and DP. I find subjectively DP to have fairly 'clean' code relative to some others I've used.

I will say as a DISCLAIMER. It's probably gonna be video taped and end up on the studio I work at's site. We have a channel in the works for all kinda of things.

But this should be fun, and subjective. We are not scientists, and our labs are all different. So are our tools. The inetersting thing is hearings everyone's opinions, because they can hear it in full res, on their own familiar system. I am not out to prove anything besides differences, because I have already formed a fairly strong opinion on the topic, after listening to many people and boxes.

To me it's what are the diffs, how important are they, and is it worth it financially. Also, it should show, how converter/box design effects boutique gear. (I'll plug the calrecs in :)) we obviously aren't comparing just conversion, but the boxes capture ability I believe, but I'm still extremely novice in electronics.

So I'm down for watever and I'm always happy to spend a few hours in the studios. It'd also be a good kick in the butt for our YouTube channel that doesn't exist yet....

All in good fun and in then name of art. Swearing, and name calling is allowed lol anywaone else.?

I think it's safe to assume that generally, we get a level of quality that is commensurate with what we pay for. There might be the odd exception to that rule; occasionally, here and there, a particular piece emerges that we are pleasantly surprised by, considering its low price - but for the most part, the higher the quality, the higher the price. This includes everything in the chain; pres, converters, mics, along with monitors, power, cable, etc.

Personally, I noticed an immediate - and substantial - improvement in fidelity when I upped my mic pre from the standard Presonus VSL I had been using, to the ADK ( not wanting to start a debate on XFO based vs transformer-less, just sayin that I noticed a big difference in quality.) I'm still using the Presonus as the i-o, so obviously I'm still using its converters. And to my ears, it doesn't sound bad at all - I think they sound good, but, I've also not had the opportunity to hear what the ADK would sound like when integrated with a higher quality conversion system. And, I've also not recorded much more than one or two tracks at the same time.

If what some people say is actually true, and there really is no difference in sonic quality between a $49 Soundblaster, a $100 Behringer mixer, and a $59 Chinese condenser mic - and that of Pro level gear, wouldn't we all choose to save thousands ( and thousands) of dollars and be using that cheaper gear, if we could get the "same quality" out of the cheaper gear - that we do out of the pro level gear that we choose to use instead?

It's not just about price with me, so that I can brag about how much I've invested. I don't choose to use AKG 414's or Neumman U87's or 89's over Behringer condensers just because they cost more, just so I cna say "I've got $6000 invested in microphones".
I use them because there's a gigantic difference in sound quality ... the pro mics that sound infinitely better to me just happen to cost a boat-load more than the Behringers do. LOL.

If I could get get the same quality in sound and build from a $99 condenser mic that I now get with a $1800 Neumann U89, believe me, I would most certainly opt for the $99 mic. Wouldn't we all? ;)

I think the only way to put aside taste, emotions and ear training and limitations, is not to listen to the end result but use some kind of forensic analyser that will compare the signal before and after going through a converter. Then take the same exact signal and do the exact same test through a second converter, then compare the results. Of course this would also admitting that we can't compare converter chip alone but only with the circuitry surounding it.

The thing with anylzers is they don't account for timbre. Timbre is the most overlooked aspect of signal integrity and room discussions. Timbre is what most people are hearing when you ask them 'how does this sound'.

I'm interested in the subjective. Can we all hear the same difference in our different spots. Does the conversion become more obvious on my iPad an mackies, than on the big guys at the studio?

I'm interested in opinions based on real world conditions. I think it will help demystify the question of 'for who is it truly worth'. It might help us better define what we need vs want.

kmetal, post: 428699, member: 37533 wrote: The thing with anylzers is they don't account for timbre. Timbre is the most overlooked aspect of signal integrity and room discussions. Timbre is what most people are hearing when you ask them 'how does this sound'.

I'm interested in the subjective. Can we all hear the same difference in our different spots. Does the conversion become more obvious on my iPad an mackies, than on the big guys at the studio?

I'm interested in opinions based on real world conditions. I think it will help demystify the question of 'for who is it truly worth'. It might help us better define what we need vs want.

Exactly Kyle, and really what this entire industry is all about. Whats good for me may not even apply to the next guy. Everything is so subjective and isn't this really why one guy solves the mystery when others are still left scratching.

Lol that's when they call someone like us right!? Fitting the right pieces for an 'overall' best in an individual's room is as trivial as it is rewarding.

kmetal, post: 428699, member: 37533 wrote: The thing with anylzers is they don't account for timbre. Timbre is the most overlooked aspect of signal integrity and room discussions. Timbre is what most people are hearing when you ask them 'how does this sound'.

I get what you are saying but we're talking about converters right ? They're is the last place I'd want to worry about modifying sounds/timbre. We forge the audio with the room, mic and placement preamps. if you're getting into also chosing the right converter for the song I'm hanging myself right away. To me converters should not transform sounds. it should replicate it with the maximum of acuracy. In a sens, If the converter is accurate, I assume that it will sound good because my signal chain sound good.

Having that in mind, to me, comparing converters would be sending the same exact signal through them and A/B the results. Now, I'm very much aware of my ear limitations and possible placebo effect. That's why I'd want a null test and spectrum analysis and I'd place the ears and human factor at the end...

Of course I admit, if a converter is not accurate, the result may sound better or worse than the signal I sent to it and this is when listening to it is important.

I'm all for bench comparisons, but unless we were all on identical setups, our subjective opinion on the converters is influenced by other factors, room speakers ect.

When you get to these high end pieces they will all test remarkably. Most of people who choose converters on this level, it's for subjective sonic performance. With the usual words like depth of field, and low end response flying.

My thing is, on my iPad I probably won't be able to hear much of a diff between motu the Rosetta and the ensemble. Would I when I hook up my mackies? Do I have to go to one of the studios before it makes an audible difference. Basically where is High end conversion most appropriate from a cost benefit standpoint, for a consumer and a professional. Ye ol point of diminishing returns. Lol

kmetal, post: 428870, member: 37533 wrote: My thing is, on my iPad I probably won't be able to hear much of a diff between motu the Rosetta and the ensemble. Would I when I hook up my mackies? Do I have to go to one of the studios before it makes an audible difference. Basically where is High end conversion most appropriate from a cost benefit standpoint, for a consumer and a professional. Ye ol point of diminishing returns. Lol

That's what I'm saying. If a person has 2 converters he must produce 2 files the exact same but with only the converter change. They he and other people can compare these tracks with forensics and by ears. The sad thing is, no one has 20 converters to make a diverse enough test that will help people decide which one to buy... Well if someone those, contact me please ;)

I own 2 converters worth testing ; the AD96 and those in the 4-710

Once you are in the high end converter market, the prices and sonic s are comparable. Higher end has includes additional options. Mid and low end usually don't have some of the options that come with example, [="http://www.rme-audio.de/en/products/adi_8_qs.php"]RME ADI-8 QS [/]="http://www.rme-audi…"]RME ADI-8 QS [/].
I've owned a few of these with [[url=http://="http://www.rme-audi…"]AES HDSPe 32 [/]="http://www.rme-audi…"]AES HDSPe 32 [/]and nothing to this day has matched their low latency response and ease of passing audio to and from a hybrid DAW. They are by far the best professional converters I have ever owned, but they also cost 4 times as much as an example: the Orion32.
Sonically, Prism, Lavry, RME, UA and Antelope with or without the 10M (no difference) are all close enough until you start using them in specific applications.

Generally speaking, If you don't need added options that you can see on example: an ADI-8 QS', then you know not to spend extra money on this in a product. Its that simple. Maybe you only need a sound card which in that case, I'm guessing you don't even have any outboard gear.

As noted by Ethan and many others who have done the round trip AB comparisons... if you round trip a digital signal a few times, (talk about ridiculous) ... your lush analog gear you bought to do all this in the first place starts sounds pretty bla.

I'd be finding ways to preserve quality rather than heading down the path of all being equal to the bottom converters of the pack lol.. Uncoupling seems to be a good way to avoid this to me.
Maybe find out why we need options over how 5 different boxes compare in sound.

Assuming what I say holds true... by the time we are actually mixing audio at this level, and have the workflow to warrant/ support better converters and the analog goodies..., the null test Ethan is pushing is imho way down the importance mark.

Comparing transparencies may seem important but I can say from my personal experience with conversion in various hybrid workflows, null testing is not what I would waste a moment on at this point of my career. All the other aspects I mention are far more important.

Its as simple as this.

I don't buy a USB converter if I am concerned about super low latency in 32 channels. I don't buy 8 channels of RME ADI-8 QS for mastering for an uncoupled capture. I don't buy an 8 channel converter with 8 pre's in it if I don't plan on using the pre's. I don't buy the 10M if I am using an internal PCIe and no other outboard converters.
I don't buy a ADI-8 QS if I don't also buy the matching PCIe interface.
I don't buy 25pin dsub tascam pinout if my console doesn't have this interconnection as well.

Once you get into real world use, you should start figuring out what you need for yourself, which is way more valuable. You get what you pay for.

just a cross reference:
http://recording.org/threads/adcs-what-is-important.53957/
http://recording.org/threads/what-is-clock-jitter.45012/

audiokid Boswell pcrecord kmetal Kurt Foster

Is clock jitter even a concern anymore? I mean, obviously it's a concern to those who are designing and building converters, but I can't really remember the last time I heard of someone using a "standalone" clocking device.

If you are using a 2 DAW system, everything I've read and heard points to actually eliminating a clock from that connection for optimal results. I think the first time I heard this was though Chris and Bos ... (I've heard it from others since) when he found out that not only could he save big money by getting rid of the clock ( because it was unnecessary) but that he was also achieving better fidelity without one.

I feel that clocking is still important and I know I get better results clocking from my AD96 or my UA-4-710 instead of from my interface saffire 56.
Now I'm using worldclock to sync 3 units in a chain. AD96 to the 4-710 to the 56. I'm guessing that in a chain mode, we could encounter problems if we add more units to the chain because of the delay the signal needs to arrive at each units.
What I don't know is how many units could we chain together before problems occur.
But I'm pretty sure that at some point having a dedicated clock becomes important for bigger setups.
Would I need one in my home studio ? NO ! ;)

DonnyThompson, post: 429275, member: 46114 wrote:
Is clock jitter even a concern anymore? I mean, obviously it's a concern to those who are designing and building converters, but I can't really remember the last time I heard of someone using a "standalone" clocking device.

If you are using a 2 DAW system, everything I've read and heard points to actually eliminating a clock from that connection for optimal results. I think the first time I heard this was though Chris and Bos ... (I've heard it from others since) when he found out that not only could he save big money by getting rid of the clock ( because it was unnecessary) but that he was also achieving better fidelity without one.

Yes, it's still a concern, but things are changing in several areas.

Firstly, acceptable quality clock design is spreading downwards from high-end gear to lower cost converter boxes (A-D and D-A) and all-in-one interfaces. However, as was brought out in the threads that Chris linked, a good clock can make a low or moderate-quality converter perform to its best, but it can't necessarily turn it into a good converter.

Secondly, the process of recording and mixing is changing, with much more being done inside a DAW, where clock quality does not have any effect. Clock quality only comes into play at the interface between the analog and digital domains.

Thirdly, the "two-box" mixdown technique expressly avoids common clocking, so the whole palaver of generating and distributing a high-quality clock around the studio is no longer needed.

I've lost the history on this, but this is one of those tests that is complicated to design and inevitably subjective in conclusion. We cannot even determine what 'better' actually is, definition wise. For years, studios recorded their material, and then joe public listened to it with a smiley face eq applied. We can now measure the data very accurately, but unless it hisses or distorts audibly, every listener is working on their own set of parameters for making judgements.

Earlier, did somebody actually say that ten year old tests are still valid? Technology is now advanced enough to mean we are talking subtle and personal.

If we split a sound source and fed it into numerous preamps, and then compared the results we'd certainly get a measurable noise figure difference, and maybe that's all?

I've spent a lot of time on a project recently that involved a timeline with a couple of flavours of mp3, and then 44.1, 48 and 96K sampling rates with bit depths between 16 and 32 floating, and because my DAW was not liking this very much, I ended up converting everything to 44.1, 16 bit for everything. After doing more than twenty different songs, I lost complete track of which original format was which. None leaped out and said mp3 or 96K wav. They all sounded similar, but just a bit different. All were perfectly usable.

If we weed out poorly designed preamps, then what we are left with is personal preference.

In the end converters wind up being a part of your sound. In my opinion the least effected by changing them if you have something decent.
If it's a question of you not liking what you hear, then try something else, just like you would do for mics, pres, room treatment and speakers.

Does it really matter? What is your benchmark for it being worth it to upgrade? Is it because you'll get a better sound on a technical level or a better sound overall? Chances are, judging from the myriad of tests I've heard, there are differences between every piece of gear, converters included. But a song is a song and unless you are comparing a Digi o1 to something cutting edge nowdays, the difference isn't that large.

Also the difference between bottom of the line to top using current modern gear is even less.

That said, I like tests and any tests you guys do would be good to hear just to be able to hear the differences between converters, however small, for whatever reason.

I have a Maudio profire 2626, Behringer ADA 8000 and Behringer ADA8200 through an RME Digiset I can compare if wanted.