I ran across this gem in another forum:

I have a friend who plays 44.1k and 96k demos with buttons clearly labeled with "44.1" and "96". Everyone who listens agrees the "96" button sounds better. Except he mislabeled the buttons... Everyone is actually picking the 44.1 sound because they think they are supposed to think that one is better.

He doesn't say whether the "demos" are 16/44.1 or in their original formats, and he doesn't say whether they are being auditioned on high-end playback systems or consumer hi-fi equipment. But it would be an interesting experiment to try, especially for you folks who participate in active local user groups where you could play the demos for a number of experienced recordists in addition to "regular folks".

ORIGINAL: daverich
I record at 88.2khz in the studio pretty much all the time now. The reason is not that the audio itself sounds any different, but that the plugins sound so much better at higher rates.

Sonitus EQ for instance gets a great top end at 88.2

I just wanted to jump in on this point from Dave... speaking of Bob Katz,

See this somewhat older thread on upsampling for mastering where a lot of this bandwidth stuff was discussed: http://forum.cakewalk.com/tm.asp?m=383093&mpage=1

In particular, look at this informative response from Steve D: http://forum.cakewalk.com/tm.asp?m=383093&mpage=1#384098

The thread discusses advantages of higher sampling rates, specifically for mastering. One of the topics that was brought up was something outlined in Katz' book -- non-linear processing should operate at higher sample rates to avoid aliasing of frequency components introduced by the processing (Mastering Audio, pg 226).

So... yes, as UnderTow says: choose your plugins carefully... then you should trust that plugin processing at 44.1kHz or 48kHz or whatever will give you the "right" result (i.e., no harmonic distortion artifacts). In short, look for the well-engineered plugins that utilize upsampling and proper filters, I think.

Interestingly, Steve D's post indicates that through his digido conversations with Bob Katz he concluded that upsampling (for mastering) only makes sense, basically, if you're doing SRC through top-end hardware converters. If that's the case -- then at mixing aren't we subject to the limitations of the SRC built into these upsampling plugins???

If we are subject to "inferior" software SRC within the plugins themselves, then one solution would be to record at the highest rate, then drive the plugins at that same native rate.

If the software SRC built into the plugs is not inferior, however, then the general rule would likely be a.) choose the right plugins, and b.) record at the target rate for the project (CD, DVD, etc.).

Sorry to jump your post, bitflipper... but to follow up a bit more on Dave's recommendation for processing plugins at higher native rates:

http://www.gearslutz.com/board/mastering-forum/119029-up-sample-not.html

With upsampling plugins you run the risk of bad SRC on the way back down to 44.1 or 48, and as one of the posters points out:

There is always a cost form the added processing of sample conversion. If you oversample and downsample just once, the penalty is small. Any processing device that oversamples has this loss. Stacking multiple oversampled processes reduces audio quality. I'm talking about the plugins/outboard that claim to process at a higer rate internally. Put many devices of this type in the chain and you get a hit in audio quality. Is the quality hit important? It depends on the project.

I think the safest approach is the one that Dave uses, unless you really trust the engineering behind your plugins. If you're bandwidth limited it's probably fine to track at 44.1, then either mix to stems and upsample the stems to 88.2 or 96 before applying non-linear plugs, or just upsample everything to 88.2 or 96. Of course, you could also just track everything at 88.2 or 96 from the get-go, but for large track counts that could be asking alot of your hardware...

For software SRC Bob Katz seems to be happy with Voxengo R8Brain Pro: http://www.gearslutz.com/board/mastering-forum/77212-r8brain-pro.html#post791777

ORIGINAL: keith

...

So... yes, as UnderTow says: choose your plugins carefully... then you should trust that plugin processing at 44.1kHz or 48kHz or whatever will give you the "right" result (i.e., no harmonic distortion artifacts). In short, look for the well-engineered plugins that utilize upsampling and proper filters, I think.

...

If the software SRC built into the plugs is not inferior, however, then the general rule would likely be a.) choose the right plugins, and b.) record at the target rate for the project (CD, DVD, etc.).

Maybe another side-track from original subject but,

what would be the best way to keep this possible issue down or tracked? Are ther some software for to measure harmonic distortion (or other possible bad behaviouring) 'on the fly'? My method has been lately (if needed) using the C. Buddes Plug-in Analyzer (standalone software), which I use for measure plugins either one or several at a time in series or/and parallel, just in same order as those are set up in project. I know it's not the best possible method since data in measures is different from the data in project and from all other settings as like gain/level settings are not equal but, since I have not found better method yet ...

Junski

I know it's not the best possible method since data in measures is different from the data in project and from all other settings as like gain/level settings are not equal but, since I have not found better method yet ...

Maybe this guy found a good test without realizing he did:

http://forum.cakewalk.com/tm.asp?m=1070975

ORIGINAL: bitflipper

Since I always record at 44.1, I'd like to pose a (maybe dumb) question to those of you who record at higher sample rates: if you bring up a 96KHz wave on a spectrum analyzer, do you actually have anything in there above 20KHz? Or does the converter band-limit everything at 20KHz regardless of the sample rate? In other words, are you actually recording ultrasonic material? (OK, so that's 3 questions, sorry)

Here maybe is an example (analog master -> DVD-A (vs CD vs LP)) but, are those analog masters processed when digitalized ... ?

Junski

Interesting pictures, thanks for the link.

There really isn't very much up there (above 20K), is there?

There aren't any numbers, as it's a spectral display, but it looks like an Audition screenshot and my experience using Audition's spectral display is that when the picture's that faint and purple that it's barely audible even when it's well under 20KHz.

Yes, that does make sense.

The big question, I guess, is does it matter? We've established that ultrasonic frequencies do combine to form valid audible components, and therefore contribute to sonic fidelity. But that combining action happens in acoustical space while recording. Once it's been converted to digital, the ultrasonic components have already done their work and we should only care about audible frequencies from that point on.

Preserving them through the processing phase might give them further opportunities to subtly contribute to the sound, but ultimately they're definitely going to be truncated prior to distribution at 16/44.1. In the meantime, they are as likely to degrade sound as enhance it. Seems to me that preservation of ultrasonics is a concern in the initial analog realm only, through the use of high-quality microphones and preamps.

And here's another consideration that hasn't been touched: jitter becomes a much bigger problem at higher sample rates. At 44.1, jitter isn't really a concern with the fairly high-quality prosumer converters most of us use - the built-in clocks are more than adequate. The people I talk to who swear by high-end wordclocks and claim noticeable improvement are all recording at high sample rates - maybe not a coincidence.

ORIGINAL: bitflipper
And here's another consideration that hasn't been touched: jitter becomes a much bigger problem at higher sample rates. At 44.1, jitter isn't really a concern with the fairly high-quality prosumer converters most of us use - the built-in clocks are more than adequate. The people I talk to who swear by high-end wordclocks and claim noticeable improvement are all recording at high sample rates - maybe not a coincidence.

Dan Lavry (of Lavry Engineering who makes top-notch converters) has said that if your sound improves when you run a converter with an external clock, then your converter is defective. Or to put it another way, there's no excuse for the circuit that accepts the incoming clock to do a better job than the circuit that generates the clock internally in the first place--best case scenario is that they should sound equivalent.

I once asked Greg Hansen (who developed the Zefiro ZA2 digital I/O card, and the Zefiro Inbox) why the Inbox didn't have a wordclock input. He said that it originally was going to have one, but they took it out since it increased the noise of the box unacceptably.

As far as high sample rates, remember that virtually all converters nowdays (since about year 2000 or so) are low-bit depth, oversampling, delta-sigma converters. Meaning that they sample at a very high rate using only a few bits (like 2 or 4 bits and 7.68 MHz sample rate) and then they sample rate convert to whatever bit depth and rate are requested. Whatever jitter there is in the clock, the distortion should be about equivalent regardless of the ultimate sample rate.

I have checked several technical references, and they are consistent in this: if you can hear a difference in sound quality at higher sample rates, it is due to inadequacies of your hardware, specifically of the anti-aliasing filter.

They are equally consistent in saying that most of the commonly-held misconceptions about the subject can be traced back to older converter designs, and that any "pro" converter manufactured over the last 10-15 years will not suffer the problems that led people to resort to higher sample rates in the first place.

if your sound improves when you run a converter with an external clock, then your converter is defective

I believe this to be true, and it makes sense. Furthermore, most folks on this forum probably don't need an external wordclock anyway, given that we typically only have the one clocked device (digital monitors don't count).

Whatever jitter there is in the clock, the distortion should be about equivalent regardless of the ultimate sample rate

This, too, makes sense, I think. On the presumption that you're really sampling at (approximately) the same rate regardless of what the final sample rate is that the converter is putting out. I am retracting my statement about jitter being a bigger problem at high sample rates, at least until I find where I read that (assuming I didn't just dream it).

My Thinking-

Higher sample rates result in better digital summing because every extra "snapshot" will result in more detail and "math" and reduced decimation.

Also, in response to recording at 44/24 if the end product will be 44/16....Won't the bit reduction kill any advantage to keeping the sample rate the same?

I Record at 48/24 to get a bit extra air and resolution during summing. 96k eats too much space and seems overkill.
I upsample to 96k for mastering because i use analog outboard and the digital Limiter/clipper sounds better at higher sample rates.

However, my sound always gets crippled when i finally export at 44/16, mostly just a loss of dynamics and high-end.

ORIGINAL: Roflcopter

I am retracting my statement about jitter being a bigger problem at high sample rates, at least until I find where I read that (assuming I didn't just dream it).

This guys sound like he knows his stuff, I cannot find an outright statement in the text saying so - but you would be inclined to think so.

http://www.jitter.de/english/how.html

The guy that wrote that article also came up with this: http://musicthing.blogspot.com/2005/11/paint-your-chips-with-gunk-for-that.html

Quote from here: http://www.altmann.haan.de/tubeolator/default.htm

"transforms transistor sound into tube sound
transforms cold and harsh sound into warm emotional sound"

Do you still trust him?

UnderTow

Can you explain what you mean by that distinction? I think I get you, but I'm not sure.

Ah, thanks bitflipper and tarsier! This was something I had experience with my old Delta 44 card. Whenever I would record with the Delta at 96 KHz I would notice a big improvement in sound quality. But now, with the Fireface 400, I can't really tell a difference between material recorded at 44.1 KHz or even at 192 KHz.