Well...if you only plan on using plug-ins for mastering, then the soundcard has no impact whatsoever on the sound.

However, for monitoring, yes you do need a decent to good soundcard.

In reality, if you're just using plugs for mastering, the caliber of the soundcard is probably not that important as you're likely to not notice a big difference between big, expensive cards like the Lynx cards or cheapos like the Creative Labs, etc.

J.

Thanks Cucco, I regard your reply, I´ve been afraid that I am wrong.
(btw: I paid visit your new studio websites – congratulation!!)
:D

I'm with Cucco to a point - But I'll add that there is no more important converter (IMO) than the converter that's feeding your monitor amp. If you have a "cheesey" sound card with cheesey converters, you aren't hearing the whole story.

From an electronic stand point, the card must be able to send and receive digital signals with accuracy and low jitter. From a software point, the card's drivers need to be written well so that data can transfered between card and application in a stable manner without degradation.

So my opinion and experience is that even though you are *only* transfering digital audio, quality does differ from card to card.

Zilla wrote: From an electronic stand point, the card must be able to send and receive digital signals with accuracy and low jitter.

There is no jitter transferred in the digital domain, all that matters is the data is bit accurate.

From a software point, the card's drivers need to be written well so that data can transfered between card and application in a stable manner without degradation.

Again, assuming pure digital transfer, "without crashing or dropping bits or changing bits" is closer to what the card needs to do.

So my opinion and experience is that even though you are *only* transfering digital audio, quality does differ from card to card.

Doubt it, unless the card is broken. Does ribbon cable quality (assuming data perfection) from your HDD inside your computer make a difference? No of course not!

DavidSpearritt wrote: There is no jitter transferred in the digital domain, all that matters is the data is bit accurate.

Which digital domain you are specifying. If you are referring to the domain which exists between the audio card and the motherboard, then you may have a point. In my post, I was referring to the domain which exists between the card and the outside world. In this scenario a digital audio clock certainly comes into play. Should that clock not be stable, or the input receiver or output driver be poorly designed, then jitter may very well be greater than one would want.

DavidSpearritt wrote: Again, assuming pure digital transfer, "without crashing or dropping bits or changing bits" is closer to what the card needs to do.

Mostly..yes...I think.. but I'm sorry, I just don't know what a "pure digital transfer" is. Maybe you would elaborate. None the less, it appears (to me) that driver code can influence sonics. For example, switching between ASIO and MME/WDM drivers for the same audio card can subtly change the sound. Maybe this has more do with the particular protocol than code quality. Even so, theoretically, there should not be any sonic difference because its still the same data, no? Yet...

DavidSpearritt wrote: Doubt it, unless the card is broken. Does ribbon cable quality (assuming data perfection) from your HDD inside your computer make a difference? No of course not!

We have found that ribbon cables can indeed make a difference. Differences which can be captured on hard drive. Differences which are perceived all the way down to the final burned CD. For some these differences would hardly be significant, and this would be a totally valid judgment. So doubt all you like, I hope you won't mind if I stand with the evidence of my own ears. Should you find yourself in Hollywood, feel free to stop by, have a cup and have a listen yourself.

Zilla wrote: Which digital domain you are specifying.

Any digital transfer, the only place the DATA is reclocked is in the DAC, every transfer before that is buffered.

Should that clock not be stable, or the input receiver or output driver be poorly designed, then ...

Its broken.

I just don't know what a "pure digital transfer" is. Maybe you would elaborate.

Any transfer of data in the digital domain.

None the less, it appears (to me) that driver code can influence sonics. For example, switching between ASIO and MME/WDM drivers for the same audio card can subtly change the sound.

This is implausible and debated many times.

Maybe this has more do with the particular protocol than code quality.

No that can't be it either, its all just data flowing along, if one bit is dropped or changed then the system is broken.

We have found that ribbon cables can indeed make a difference. Differences which can be captured on hard drive. Differences which are perceived all the way down to the final burned CD.

Big sigh here.

So doubt all you like, I hope you won't mind if I stand with the evidence of my own ears.

Of course not, I respect that you hear a difference, but you are listening to a broken system.

Should you find yourself in Hollywood, feel free to stop by, have a cup and have a listen yourself.

Thanks Scott, I would like to meet you and listen to these anomolies. :)

None the less, it appears (to me) that driver code can influence sonics. For example, switching between ASIO and MME/WDM drivers for the same audio card can subtly change the sound.

This is implausible and debated many times.

Maybe this has more do with the particular protocol than code quality.

No that can't be it either, its all just data flowing along, if one bit is dropped or changed then the system is broken.

It seems as though this should be VERY easy to test.

Send the same signal into the workstation digitally. One using one driver type, the other using any other variety.

If there is in fact a difference a simple phase reversal would reveal the differences.

Anyone game (and have the extra 10 minutes to try it?)

Mr. Spearritt - I think you know I'm always game for tests like this. Set up the parameters and I'll run it accordingly.

J.

Yes, its very easy to test. I have done it, ie ASIO vs WDM (or should that be WMD) and the files are identical as expected.

As I always say, if digital transfers to the final resting place in the DAC, contain different streams from the same source, then the system is broken and needs repair not debate!

I'm in complete agreement with you. Perhaps a test would be good for the forum though...

Theoretically this is true, but in the real world I don't find this to be true. I can run the same audio on the same computer from the same hard drive through the same DAC using only different software and it sounds different. I don't see my system as being broke, but I do hear a difference. there can be many things that cause this difference and I can spend countless hours, days, months trying to find out why. OR I can just go with the better sounding one. It's not a subtle difference either. I can run say protools in OS9, play the same audio in OSX and it's different, then play it in logic, different yet again. I'm not tech geek so I can't tell you exactly why although I know enough to paint myself into a corner. But I've got things to do and sessions to complete so I go with what sounds the best to me.

Michael Fossenkemper wrote: Theoretically this is true, but in the real world I don't find this to be true.

That's because people's idea of the real world, is to conduct flawed experiments, like ...

I can run the same audio on the same computer from the same hard drive through the same DAC using only different software and it sounds different.

Careful testing is required with software comparisons, checking that no unintended dithering or plugins or gain settings are active. Firstly, very basic testing is required, simple WAV file playback, with the digital output just before the DAC being recorded and data compared, then one works from there.

I don't see my system as being broke, but I do hear a difference. there can be many things that cause this difference and I can spend countless hours, days, months trying to find out why.

Or, you could try to get to the bottom of it with some simple tests, actually there aren't "many things that cause this difference" at all.

OR I can just go with the better sounding one. It's not a subtle difference either. I can run say protools in OS9, play the same audio in OSX and it's different, then play it in logic, different yet again. I'm not tech geek so I can't tell you exactly why although I know enough to paint myself into a corner. But I've got things to do and sessions to complete so I go with what sounds the best to me.

Well, this is a sensible way to go, but you then can't really say one sounds better than the other until you know why, to eliminate any faulty experimental technique. As there is, clearly, no consensus amongst industry professionals about the best sounding software, this is proof positive that flawed experiments generally drive these conclusions.

Well I've been around the block for awhile and like to think I know a little bit about what i'm doing though I don't really spend time conducting conclusive experiments as to why software sounds different. But when there are a few vet engineers sitting in a room and we all turn and look at each other when we hear a difference... now most of these differences I hear are when you have the audio engine perform some kind of calculation. I'm guessing it has to do with how and where the calculation is being performed, but I don't know. Now when I say "best one" of coarse that's my subjective opinion and is any thing processing wise in audio. are you saying that all audio software or engines perform the exact same calculations in the same way?

Michael Fossenkemper wrote: are you saying that all audio software or engines perform the exact same calculations in the same way?

That is the argument in fact.

David and I have definitely voiced our opinions on this subject numerous times in the past(opposite opinions that is). We even did a mild test, which, arguably proved his point. However, my thought on the subject (and on that particular test) is that it's not simply how an audio engine *PLAYS* the file, but what happens when calculations are performed on it.

Personally, I have heard significant differences between files with computations performed (even as small of a computation as a simple hard pan then bounce - using no dither).

Since I no longer have Cubase or any other DAW than Sequoia, I certainly can't put my money where my mouth is, but my ears convince me enough that I don't need to "prove" it to myself. I'm content with my decisions. I think that's what it comes down to.

The quality of every component in the computer is related in some way to
the overall audio performance. The sound card especially so,
because depending on how well it´s drivers are written will determine
how well it will function with all of the software. So you want the highest
quality audio interface/soundcard as possible with the best possible track record of compatability and updating it´s drivers for the audio programs
you use. Also, has been already said, with as good converters as
possible to play back the audio. Some good soundcards are made by
Lynx and RME (and Egosys, more popular in Asia). Lynx probably
has better converters, while RME may have better or more complete
drivers. Top of the line Egosys is high quality, low cost, not great technical
support.

There's really no doubt that soundcards make a difference. Especially when it comes to conversion (A to D or D to A).

However, if a purely digital transfer is involved, driver differences should make NO difference. The fact is, you're receiving a straight binary stream. Either it receives it (and writes it to disk) or it doesn't. If it receives it and you don't have a 1 for 1 match, something IS wrong.

In otherwords, the sound card is either *dropping* bits or modifying bits. If it does either of these...sell your card on Ebay.

A guy I used to work with used to complain endlessly about "how bad Sony ACID sounded" compared to our other programs. He'd complain about this on an almost daily basis. We get project files from people in all different formats from people who work in their home project studios, so we run about 5 different DAW programs.... and he claimed to hear differences in one or two of the programs. Usually his complaining was confined to ACID, but occasionally he'd complain about the "sound" of other programs.

Personally being the EE type, I always thought my friend was crazy and just hearing things.... My hypothesis at the time was: I think what he was hearing was the pricetag difference between one DAW program and another.

Still I never dismissed his opinion entirely since I was never truly familiar with how the software works internally. Intuitively, youd think all drivers and DAWs are doing the identical thing (more or less)... maybe not. My other hypothesis was that he had some setting wrong in ACID and was maybe truncating files instead of dithering when he rendered them.... but he'd complain about the sound even when the files were not rendered yet.

I don't want to start another long thread about this as its all been done before, many times, eg. here is a classic thread...

http://

1. Michael, yes, this is what I am saying, all the big name DAW's now use the same calculation routines, ie accepted best practice methods. If one was noticeably better than another, we and the other software teams would know about it.

2. I am assuming that we are using a high quality external jitter immune DAC, and that no domain changing is occurring in the sound card, that is a can of worms and obviously one will sound better than another depending on the differences in conversion, this is not the argument here.

3. Aracu, the "quality" in a purely digital system is less important than where a domain changes and completely irrelevant if data is transferred correctly, which is the whole reason we went digital in the first place, ie to eliminate variations in sound. We are talking a purely digital system. My Nagra V digital recorder is no better than an M box if we are both using an external digital source. No better, as long as the bits are recorded without dropping or changing any.

With respect ...

So If i'm playing a file in a "system" and the only thing that changes is the software playing back the file, then I'm just hearing things. I find that kind of hard to believe, unless you are just talking about the major players in the last few years. I don't believe protools 5.0 is using the same coding, or whatever it's called, as 7.0. I can believe that in the last couple of years the better stuff gets around a eventually makes it's way into all of the platforms.

also what does jitter have to do with it if the audio is being played in the same system?

Here is a possibility that may not have been thought of. I coincidentally stumbled upon an article today which shows that this could, at least be partly responsible in some situations.

It was an article on the various stereo panning laws, and as many of us know, the various softwares allow you to adjust this to your exact specs, or at least, choose from a default menu of the most common.

Now, normally, the "default" is -3dB sin/cos panning..... This is kindof the defacto universal standard... but some people like to tweak it. One thing I have noticed is that on my various DAW programs... sometimes this gets changed inadvertently by somebody... and sometimes when you get a new program it may not even be -3dB by default but 0dB instead. I just reinstalled SONAR today because my old one was crapping out, and I noticed it was on 0dB sin/cos (not sqr rt) law by default instead of -3dB.

The panning law used changes the sound quite a bit... its not like its a piss ant insignificant difference which is barely noticeable. So... that could be one theory which not only is technically sound... but practically speaking... its a possibility also. If the default setting gets changed in the menu... its going to change the way the stereo field is perceived.

Michael Fossenkemper wrote: So If i'm playing a file in a "system" and the only thing that changes is the software playing back the file, then I'm just hearing things.

No, you might not be hearing things, but you are hearing something that can be explained simply, eg. one DAW applies dither by default on playback which you are unaware of or one of the DAW's is truncating 32 bit files to 24 on playback which you were unaware of. In essence you are not comparing apples with apples (pardon pun).

All you have to do, is if you hear a consistent difference you need to record the data stream going into your external DAC and compare the data. This will explain any real differences. If the data is the same, bit perfect, then you are just hearing things.

I find that kind of hard to believe, unless you are just talking about the major players in the last few years.

Yes, I am talking the current crop which are about 5-7 years old, Sequoia, Samplitude, Wavelab since v4, when 32 bit float compilers and code were used in the engines.

I don't believe protools 5.0 is using the same coding, or whatever it's called, as 7.0. I can believe that in the last couple of years the better stuff gets around a eventually makes it's way into all of the platforms.

You could try to find out though. Some of the PT systems were not code based engines but hardware based, I am not referring to those.

It would become clear pretty quickly if one DAW sounded consistently better than another. So many people have been doing nothing else but listening to converters and DAW's in the past few years. There has been huge piles of debate about this on the net. It seems most engineers have been listening to converters and DAC's more than music, there is so much angst about it all.

The software writers are all using the same compilers, ie state of the art, to write these engines, they are all just number calculators.

Also what does jitter have to do with it if the audio is being played in the same system?

Bad jitter susceptible DAC's were the only reason people were hearing differences when playing different burn speed CD's or listening to DAW's a few years ago. Apart from the conversion quality itself, the reclocking of data in a normal CD player or cheap DAC has always been where most of sound quality and differences come from. It just removes this variable from a DAW comparison experiment where the DAW has nothing to do with any D/A anomalies.

I should be clearer about what I mean by ¨quality¨ in referance
to using audio interfaces/soundcards that have well written
drivers. For example, if I´m using a soundcard that is
not very compatible with a particular program because that
program uses wdm drivers but the soundcard´s wdm driver
is not well written, the computer might crash occasionally
when trying to use that program, which will effect the audio
¨quality¨ by completely interupting the process. Or certain
aspects of the program won´t work at all etc. Thinking in a
similar way, if the computer is not well ventilated and heats
up too much, it can cause the computer to crash, or hard
drives or other parts to wear out, which will effect recordings
or whatever the work process is.

I agree that the choice of program to a certain extent has more to do with how and what you are using in conjunction with it. I also agree that there is probably too much time spent on figuring out which one is "best". Like I said before, I've got too many projects that I need to get done and I'm going to go with the one that fits my workflow the best. I'd much rather debate over things like dither and CD burners....... NOT.

It would be interesting though to see where the software side of things go. I've always thought it would be cool to have an open platform kind of system where you could pick and choose aspects of different programs and totally customize a workflow based on the persons preference. Now that would be cool. But economics probably wouldn't allow something like that to come about. Oh well.

It seems we've gone from talking about the (possible) differences between ribbon cables and digital sound cards to the way software engines process a bunch of complicated tasks. BIG difference, abliet both could still be subjective.

The only thing I agree with is the possibility of different software engines yeilding different results. There are lots of folks who say they hear a big difference between, say...Sequioa mixes and Pro Tools mixes. Is it real? Is it provable? No idea; I too don't have the time for that, and I'm damn glad I work with Sequioa anyway.

The few things I can think of that might cause a difference would be the result of dithering, bit depth, sample rate, plug in qualities, summing buses, etc., but beyond that; digital is digital.

I'm with David all the way on this, and I've been wondering what exactly it IS that everyone else is always going on about. I've been digital since 1994, and with proper care and attention to what's going on, there's no problem at all. Concern over jitter and external clock rate is important if you're slaving a lot of diverse gear together (ProTools rigs running Time code and big movie scores, lock to video playback, etc.), but beyond that, I'm always working from ONE source, ONE master clock, inside of ONE computer. It works, it sounds great, it's repeatable, and the results are what they are, time after time after time.

"Back in the day", bad digital was caused by a host of things, but the gear of today has all but eliminated that. No offense to our esteemed colleagues and professionals here, as well as the rest who say they hear something, but my response is: "oh, REALLY??? Exactly WHAT is it you think you're hearing?" The responses seem to fade out and never materialize after that. I have NEVER gotten a straight answer or a definitive, scientific, provable, hard-fact example of these kinds of claims.

So many "Tests," if conducted at all, are flawed and unscientific, and so many "Differences" are simply nothing more than level changes between the two examples. The human ear has less than a 2 second retention span, so unless someone is toggling between two IDENTICAL sources, with all other things being equal, all bets are off. Don't even THINK about telling me something you heard in one room one day sounds different than something you've heard elsewhere a day later. Uh-uh. (Whatever happened to the scientific method?)

As for all these "differences," I say: Put up or shut up. SHOW ME. Give me concrete examples. Demonstrate exactly what it is you're hearing that's oh-so-horrible, wrong, or iust bad. Give me a printout, give me a chart, give me two wav files that I can put on the timeline, toggle between each, then flip the phase on one and hear the difference when we sum them. Otherwise, it's bunk.

As for the original question: Getting the sound into the computer/digital domain in the first place is key: Good talent, mics, room, pre's and signal path right up to the converters is where all the important (analog to digital) stuff happens. Yes, a bad (old) soundcard with an all-in-one line in/ADC can sound quite bad (and noisy). But once you're inside the box, the only variable is then your software, your own mixing skills (and ears).

Sure, a $1200 DCA headphone amp is great and ideal, but there's a variety of mid-level stuff out there that also sounds good (read: Usuable), and you'll have a good example of your work in progress. (But keep it mind, it will NOT change what's going on in the wav file itself; only the software manipulation does that.) As for your playback speakers; good, bad or in between, they are just letting you hear the audio, but they do NOT change the audio while you're simply monitoring with them.

Hard drives, controller cards, ribbon cables, CPU speed, etc. all have NOTHING to do with the sound you're working on. Data flow is data; and unless someting is dropping it or corrupting it (ie: BROKEN), your results will be the same no matter what computer (or ribbon cable) you are using.

I record live to-PC with one system, and store it all on a portable hard drive. I then move that hard drive to another system in my editing suite, transferring the very same data INTO the computer (via firewire, USB 2, etc) to the internal hard drives. When the project is mixed and completed, I'll move the data back to yet ANOTHER hard drive, to go into offline storage. If you think for ONE SECOND there's the tiniest bit of difference in the data integrity or SOUND, (caused by what - platter speed? EIDE vs. SATA drives? Intel chips vs. AMD chips? Ribbon cable manufacturer????), then I can't help you.

As any of these projects make it to CD or DVD-A, then the ONLY thing coming out to the analog world is the playback device itself - far removed from the process itself. THAT is where things can sound bad, and that can be flawed, but beyond that, it's really time we exploded so many of these myths and get back to work.

JoeH wrote: If you think for ONE SECOND there's the tiniest bit of difference in the data integrity or SOUND, (caused by what - platter speed? EIDE vs. SATA drives? Intel chips vs. AMD chips? Ribbon cable manufacturer????), then I can't help you.
.

There was a discussion over at Gearslutz where many were mocking a famous producer/engineer who swore that he could hear the difference in sound between internal versus Firewire hard drives. Mmm-kay. Suuure.

There was a discussion over at Gearslutz where many were mocking a famous producer/engineer who swore that he could hear the difference in sound between internal versus Firewire hard drives. Mmm-kay. Suuure.

Excellent example, Mises. Where's the proof? Where's the track? (Before and after, A/B comparisons). When can we hear it?

I've done this test with Sequoia, Nuendo, Cubase, and Logic - no difference either audibly or with phase cancellation tests. Lynn Fuston did just this with these DAWs and others as well (in his results, Fairlight and PT HD also cancelled and sounded identical in unity gain mixes, not just single stereo files).

If there's a difference between simple stereo files on playback, then there is either something else in the path, or visual aesthetics are more of a factor than most would like to admit.

I am convinced after several years of this debate, hearing other engineers talk about noticeable differences between the same DAWs I've tested extensively and carefully, and repeating this test numerous times, that ears aren't the only factor influencing what people "hear".

Ok, i'm going to conduct a test. weeeeee

I'm going to take a soundfile 24bit, play it back once in protools OS9 5.1 whatever it is, altering the gain by 3db output through my dac and into say a masterlink. do the same thing in OSX protools, and then do the same thing in logic. I will post these 3 files. Everything will be the same except the software playing back the files, no dither will be applied. I will be capturing the analog output back to digital since this will represent more of how we generally listen (from a dac). anyone want to make a suggestion on this test or see something flawed with it?

Michael Fossenkemper wrote: Ok, i'm going to conduct a test. weeeeee

I'm going to take a soundfile 24bit, play it back once in protools OS9 5.1 whatever it is, altering the gain by 3db output through my dac and into say a masterlink. do the same thing in OSX protools, and then do the same thing in logic. I will post these 3 files. Everything will be the same except the software playing back the files, no dither will be applied. I will be capturing the analog output back to digital since this will represent more of how we generally listen (from a dac). anyone want to make a suggestion on this test or see something flawed with it?

This is just my laymans understanding of it.... but doesn't the audio get dithered (or truncated) whether you intentionally apply it or not when you output to a DAC?

Different DAW programs use different sized engines for the calculations (i.e. 32 bit float, 64 bit, etc..)... However the DAC only accepts 24 bit wordlengths.

The DAW "transparently" (without most users realizing) converts the higher wordlength (32 float or 64 bit) into 24 bit when you send it to the DAC for reference monitoring purposes.

There was a big discussion about this several years ago on ProSoundWeb in which Dan Lavry and Nika Aldrich were involved when they were arguing with others about dithering and wodlength reduction where and when it gets applied.

The only time wordlength reduction doesn't get applied is when you directly export a 32 bit float file and the software natively uses 32 bit float processing.

However, if the software natively uses 64 bit, it still gets converted to 32 float as a minimum (I think).. and when it goes through a DAC though, it has to get wordlength reduced from either 64 or 32 bit down to 24 bit.

Even if the software dithers or truncates it without you [yourself[ intentionally applying it... it still is a good test... because its still proving which DAW is doing what when it sends it out to the DAC... so even though it is doing the dithering transparently without people realizing it.... maybe it is this process which could partially be responsible for the alleged sound differences between software.

I'm still want to reiterate my previous stereo panning law theory though as a possible explanation. Ochams razor. Look for simple explanations before we get into more complicated ones.

To do this test so that its reasonably scientifically valid... all other things have to be equal. All the important little settings in the various DAW programs have to be set the same... other than, perhaps, the engine size, which is selectable in some DAWs. You should set the engines so they operate in their maximum mode.

I agree with Mises, not sure you will learn anything from the proposed test.

Michael Fossenkemper wrote: I'm going to take a soundfile 24bit, play it back once in protools OS9 5.1 whatever it is, altering the gain by 3db output through my dac and into say a masterlink. do the same thing in OSX protools, and then do the same thing in logic. I will post these 3 files. Everything will be the same except the software playing back the files, no dither will be applied.

Except that the 32 bit files automatically created as soon as gain is applied may be truncated on output without controlled dither.

I will be capturing the analog output back to digital since this will represent more of how we generally listen (from a dac).

No, this is pointless. To test the DAW and only the DAW one needs to remove spurious variables that corrupt the results from the DAW, like an unnecessary D/A and then another A/D conversion.

Please capture and compare the signal before the DAC.

Anyone want to make a suggestion on this test or see something flawed with it?

:)

If everything is working properly, the only significant differences
directly effecting the audio quality will be in ¨destructive¨
processes, especially, different eqs, reverbs, dynamics
processers etc. Other than that, the different platforms (and
soundcards also to a certain extent) will produce different
workflows, which only indirectly effects the final mix, although
not literally effecting the audio quality, unless something is
causing files to become corrupted.

I'm not sure how this would be pointless. All I would be changing is the software that is doing the calculating. Coming out of a DAC is the way we hear it, so isn't that part of the "system". By eliminating the DAC, aren't you eliminating an important part of the process? sure there will be some difference in that part of the process is going to be analog, but how else would one Capture the difference, if there would be one, in what one is hearing without including the DAC? Maybe all it would test is how it communicates with the DAC, but isn't that important also?

I'm also not sure how 32 bit comes into play being that 2 of the processes are using TDM and that is running at 24 bit. Not sure if logic bumps it to 32 bit and then truncates it to 24 bit running through core audio and then into TDM. Or maybe the very nature of core audio in OSX is running at 32 bit and then gets truncated to 24?

Hey Mike -

I don't know that, if I were you, I'd even want to waste my time with the test. It appears that most everyone here agrees on the basic concept that digital is digital, from there, all other points are quite minute.

I say, if someone wants to argue with me that their DAW sounds better or their hard drive or RAM sound better...I'll gladly concede. Then...while they're spending all their money on high end Hard Discs, funky ribbon cables and mic/line cables made from the pubic hairs of male albino unicorns, I'll spend my money on more Schoeps, better converters and maybe a prostitute or two.

I guess my point is...even though there probably IS in fact something to prove, that something is so minute, I doubt that the results would be in anyway Earth shattering.

Cheers -

J.

I agree if anything is different, it's not going to be big. And like I said before, I've got much more important things to worry about . But I may do it anyway just to see.

The original question was about soundcards, which vary in some
respects, even if not used for any analog function. I think the thread
got confused with the physics of digital audio. Electronic gadgets are
not as perfect as the physics they are based on, the physics being
different from many of the the social and economical aspects of
recording. Depending on the audio project you are working on, it
could be more useful to have a top of the line soundcard than
a Schoeps microphone!

I disagree with this conclusion which is why I posted in the first place. To answer the original question in a more direct way ... if you restrict to digital activity only on your PC sound card, and this is the sensible (best) approach, then any card which has 100% data integrity will suffice and be as good as any other digital card.

More money, generally, buys better domain conversion, A/D, D/A, but I maintain that these are things that should not occur on a PC soundcard in a well designed system.

The physics of digital audio is true in the context of a belief system.
In practice, sound quality, audio performance, software and
hardware compatability, workflow and connectability are not
separate issues. Sometimes computers produce no sound or a
distorted one where nothing is broken or misconfigured, due to incompatabilities, since many different individuals and companies
are developing products independently from one another, although
based on the same theoretical schemes. In some cases each tech
support representative will blame the other product. To minimize
that type of situation it helps to choose the computer componants
carefully.

Bored now...

Take 2 capsules of LSD and get some rest.

Sometimes computers produce no sound or a
distorted one where nothing is broken or misconfigured,

This makes no sense; if distortion has not been specifically called for or dialed in, then something IS broken, misconfigured or simply not working properly.

Computers are literal devices, (this is why we like them) they do not deviate; they exactly what you tell them to do, repeatably, over and over again, unless you tell them otherwise. Any change in that process is either the result of a defect, or a deliberately pre-programmed randomizer process, or what have you.

I think 99% of these so-called, perceived "Differences" up in that last 95-100-percentile are due to missed variables: room temp & humidity, head colds, emotional stress (or lack of), euphoria over new gear purchases, depression over late client's checks, the position of the sun, moon & stars at the time of testing, and about 1000 other intangibles. Do a double-blind, scientific test on some of these "theories", and most of it goes right out the window.

I'm hearing the lonely, far-off sound of crickets and tumbleweeds here, which is what usually happens when this type of discussion runs out of steam.

Still waiting......and waiting....for proof......... :roll: