what is the difference between non-oversampling and oversampling in a dac?

Jul 12, 2005 at 7:43 PM Post #16 of 48
Quote:

Originally Posted by Leporello
If upsampling works the way that it is suggested to work in this thread, it very much seems that we are able to create something of nothing.


I guess now I know what you were trying to say: that oversampling tries to (re)create new information that wasn't on the CD. No, that's not what it does. Rather the steep low-pass filter introduces something that wasn't on the CD: high-frequency ringing. And exactly that's what some of the upsampling algorithms try to reduce without the treble roll-off of the filterless designs or Wadia's Spline filter. IIRC the Meitner approach is one of the most complicated and interesting: depending on the music signal, it changes from time-optimization to amplitude-optimization.

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Jul 12, 2005 at 10:37 PM Post #17 of 48
Quote:

Originally Posted by Glassman
in case of ordinary CDs, you have passband from 0-22.05kHz and exactly the same spectrum mirrored from 22.05-44.1kHz.. luckily, today's amplifiers and speakers can usually take these ultrasonic frequencies without going mad.. that wasn't possible somewhere back in 80's..


I'm afraid I got totally lost in your post... but are you talking about reproducing frequencies beyond the range of human hearing or removing them?

See ya
Steve
 
Jul 12, 2005 at 10:51 PM Post #18 of 48
By the way, Blessings... I just noticed your link to the 6 Moon website while trying to figure out what people are talking about here... That site is hilarious. I really like their review of the "magic chip" you lay on top of your CD to grant it magic sound. Each chip only works 10 times, then you have to return it to get a recharged one!

http://www.6moons.com/industryfeatures/chip/chip.html

See ya
Steve
 
Jul 13, 2005 at 8:02 AM Post #19 of 48
Quote:

Originally Posted by bigshot
I'm afraid I got totally lost in your post... but are you talking about reproducing frequencies beyond the range of human hearing or removing them?

See ya
Steve



I'm explaining what happens when you play a CD on NOS filterless DAC.. in such a case you really are not removing the mirrored spectrum and so it just remains there, it is reproduced.. this is unwanted effect but NOS advocates think it hurts less than having filters engaged.. and I was hinting about why didn't NOS systems exist in the 80's when CD was born, although they were using exactly the same DAC chips most of today's NOS DACs use.. it's because back then, amplifiers couldn't take frequencies nearly 50kHz at full amplitude without problems..
 
Jul 13, 2005 at 6:31 PM Post #20 of 48
What's wrong with simply filtering out the unwanted frequencies? That's considerably easier than the roundabout resampling you describe. Or am I still confused?

See ya
Steve
 
Jul 13, 2005 at 6:49 PM Post #21 of 48
Quote:

Originally Posted by bigshot
What's wrong with simply filtering out the unwanted frequencies? That's considerably easier than the roundabout resampling you describe. Or am I still confused?


I think it's been explained ad nauseam how upsampling serves for low-pass filtering.

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Jul 13, 2005 at 8:39 PM Post #22 of 48
Well I apologize profusely for boring you with remedial topics, Mr Science! Feel free to post somewhere else if this doesn't interest you.

But I would still like to find out, why resample when you can build a simple low pass filter into the preamp of the player?

See ya
Steve
 
Jul 13, 2005 at 8:39 PM Post #23 of 48
Quote:

Originally Posted by bigshot
What's wrong with simply filtering out the unwanted frequencies? That's considerably easier than the roundabout resampling you describe.


Because of 44.1 kHz sampling, the first alias can extend down to 22.05 kHz. That means you have to have a filter that goes to -96 dB in about 2 kHz if you want to have response up to 20 kHz. That's a very steep filter, and very difficult to do well with analog filters. Oversampling/upsampling filters do this brickwall filter digitally where it's much easier, and then resample to a higher sampling rate, so the aliases occur much higher in frequency, and can be filtered out with simpler analog filters.

--Andre
 
Jul 13, 2005 at 8:46 PM Post #24 of 48
Quote:

Originally Posted by bigshot
Well I apologize profusely for boring you with remedial topics, Mr Science! Feel free to post somewhere else if this doesn't interest you.

But I would still like to find out, why resample when you can build a simple low pass filter into the preamp of the player?



As AndreYew has explained (just like myself repeatedly a few posts above), the «simple low-pass filter» is simple not before the digital pre-filtering done by means of upsampling.

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Jul 14, 2005 at 10:29 PM Post #25 of 48
Quote:

Originally Posted by AndreYew
Because of 44.1 kHz sampling, the first alias can extend down to 22.05 kHz. That means you have to have a filter that goes to -96 dB in about 2 kHz if you want to have response up to 20 kHz. That's a very steep filter, and very difficult to do well with analog filters. Oversampling/upsampling filters do this brickwall filter digitally where it's much easier, and then resample to a higher sampling rate, so the aliases occur much higher in frequency, and can be filtered out with simpler analog filters.


Thanks! I got it now. The upsampling isn't doing anything to improve the sound quality, it's just a technical workaround dictated by the format. I had it confused with the upsampling feature on my DVD player where it bumps the sound up, dithers it and theoretically sounds better (but doesn't).

Thanks for taking the time to explain it to me.

See ya
Steve
 
Jul 14, 2005 at 11:08 PM Post #26 of 48
Quote:

Originally Posted by bigshot
Thanks!


You're welcome.


Quote:

I got it now.


I'm not so sure.


Quote:

The upsampling isn't doing anything to improve the sound quality, it's just a technical workaround dictated by the format. I had it confused with the upsampling feature on my DVD player where it bumps the sound up, dithers it and theoretically sounds better...


Upsampling has always a low-pass function. Your Philips player should benefit from exactly the same mechanism as other upsampling DACs, including the simpler analog filter.


Quote:

...but doesn't.


So this technological sophistication is wasted on you...
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Jul 15, 2005 at 4:14 AM Post #27 of 48
Quote:

Originally Posted by bigshot
Thanks! I got it now. The upsampling isn't doing anything to improve the sound quality, it's just a technical workaround dictated by the format. I had it confused with the upsampling feature on my DVD player where it bumps the sound up, dithers it and theoretically sounds better (but doesn't).


It may actually affect sound quality, because of other external factors. The filters in the upsampling chip may be more accurate (or worse) than the OS filter, or the upsampling chip may reduce (or increase) incoming jitter: a jittered signal can be thought of as a signal whose sampling rate changes from moment to moment, so a bunch of companies like Benchmark and Bel Canto use asynchronous sample-rate converters to upsample whatever is coming in to a well-defined, solid higher sampling rate. Or it may be easier to transmit and receive higher sample rates with less jitter (Meridian claims this is why they do it) with S/PDIF.

--Andre
 
Jul 15, 2005 at 7:40 AM Post #28 of 48
What does jitter sound like? Is it like that helicopter sound you sometimes hear with compression artifacting?

See ya
Steve
 
Jul 15, 2005 at 7:55 AM Post #29 of 48
Jitter makes the sound less focused. Basically it's not as clear/detailed.
 
Jul 15, 2005 at 8:23 PM Post #30 of 48
Are there any blind A/B listening tests where people were able to hear it, or is it theoretical? Do SACDs have jitter?

Thanks
Steve
 

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