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).

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

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: bitflipper

If higher sampling rates don't affect jitter then why is it possible to hear a better quality recording from prosumer cards at higher sampling rates?

This is what I was responding to in my previous post. It's because of the analog filters, not the sample rate.

I would suggest that it is probably because of the digital anti-aliasing filters, not the analog filters or sample rate. Again, the converter is doing all the conversion at a very high sample rate, and thus there is only one analog filter before A/D conversion regardless of ultimate bit depth and sample rate destination. (and again, this really does depend on a particular converter's implementation. But it's a fairly common modern implementation) The sample rate conversion from the low bit/high sample rate (4 bit, 7.68 MHz for example) to high bit/low sample rate (24 bit 96 kHz) is a digital filtering process, and it can be poor quality or it can be great quality.

It's entirely possible that the above mentioned prosumer card's sound quality improvement at 24/96 is mostly due to an inferior digital filter when going to 24/48. Not necessarily because of jitter or the analog filter.

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

ORIGINAL: jon busticle

My Thinking-

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

But that isn't how digital audio works. You increase the bandwidth, not the resolution.

UnderTow

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

Do you still trust him?

Seriously, cannot say - just went over his other material, and the man sure knows his stuff, however crazy the above may sound:

maybe read this review as well:

http://www.enjoythemusic.com/magazine/viewpoint/0904/aachapter60.htm

and to stay on-topic:

http://www.mother-of-tone.com/cd.htm

scroll to the bottom, the last bit is interesting.

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.

My Thinking-

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

Don't feel bad, this used to be my thinking as well, based on what I'd learned about A/D converters in electronics school back in the early 70's.

Back then, what we saw were 8-bit R-2R ladders, used primarily for data acquisition and telephony multiplexers. Audio recording in digital was an exotic new thing back then, something you did with ultra-expensive custom-built gear. Things have changed a lot since then, as I discovered when I later revisited the topic in an attempt to understand converters better.

I won't attempt to explain why your logic is incorrect because I'm just not that good at explaining stuff, and frankly some of the math is over my head.

I can, however, recommend one source that did a pretty good job of explaining the Nyquist magic theorem -- a book by Nika Aldrich called "Digital Audio Explained for the Audio Engineer". Parts of it are mathematically intimidating, but he does a good job of illustrating how the audio is reconstructed from as few as two samples per cycle -- something that is not intuitive at all!

Until you have a chance to read the book, here's the bottom line, which you'll have to take on faith for now: Nyquist says that you can encode and subsequently recreate ANY waveform EXACTLY as long as you sample it slightly over twice the highest frequency you need to record.

This is hard to comprehend at first, because it's non-intuitive. The operative word here is EXACTLY -- not a reasonable approximation, but an exact image of the original waveform. No kidding. And with as few as two samples per cycle.