Up, over, and resampling


I tried asking this elsewhere and got a few anecdotes (more are always welcome), but I was really hoping for some links to technical papers (that I may not fully grasp. I haven’t kept up my college calculus.)

Does anyone have links to educational articles or papers on upsampling, oversampling, or resampling? I know there’s no new information generated, but it still theoretically helps DAC performance, right? Is there a downside as long as the implementation is solid? And what qualifies as correct?

I had a CD player many years ago that had an upsampling add on board. Impossible to quickly A/B, but it certainly seemed to enhance the experience enough that I had zero regrets with the upgrade. I think it converted redbook CD to 24/192. So I’ve had a positive early experience, But I see NOS DACs are crowed about, while there are others embracing a more is better approach  

I’m trying to research a little because I finally got around to reading some of the manual for my new streamer. (So I don’t screw up ripping all my CDs for, hopefully, the last time.) I realized it has options for resampling during playback. If I’m using the USB output it always sends at 32bit but can also goose PCM up to 8x (384kHz max). S/PIDF output is bumped to a maximum of 24/192. I don’t hear a big obvious difference, but I haven’t spent much time looking for one either. Without any DSP down the chain is resampling really worth it? And if I were to some day splurge and get an M Scaler or similar would it make sense to disable any resampling by my streamer in favor of a presumably better dedicated component?

Because it will be asked. My virtual system is mostly current. CD player was an Ah! Njoe Tjoeb, that was mothballed years ago when the transport started getting twitchy. The streamer is an AURALiC Aries G1.1. DACs are Chord Qutest or Soekris dac1421.

cat_doorman

Showing 3 responses by erik_squires

https://www.stereophile.com/asweseeit/344/index.html

Whoops my bad. I only got Oversampling half way correct. It does NOT interpolate, that part I was correct about, what I missed is that the intermediate samples are all at 0.

Also, I agree with the following statement, that the best thing these filters do is slightly tweak the top octaves:

So while I strongly suspected that the improvement I heard with the dCS 972 was simply due to its using a different oversampling filter, along with the benefit of better downstream DAC behavior when fed a 24-bit rather than a 16-bit signal—as I described in my January 1996 review of the Meridian 518—I wasn't sufficiently sure to spill ink on the subject.

The oversampling is usually a multiple of the native rate 44.1kHz can become 88.2, 176.4, or even 352.8kHz which is 8x over USB. (Similarly 48kHz can become 96,192, or a max of 384kHz.) At its crudest existing value repetition could be used,

 

OP: Well wish you had read my post. This is the ONLY definition of OVERsampling there is. The original values are repeated. This requires no processing but improves the behavior of the output filters. If it isn’t doing exactly this it is something else.

The crudest, best and average oversampling systems all do this. The only variable is the number of oversamples per original.

It hasn't helped anything in 20 years.  It may have a modest effect on the top octave of the DAC's frequency response.

Imagine two data points on a musical signal on one channel.  The first point is 40, the next point is 48.  The raw data:

40, 48

4x Oversampling:

40, 40, 40, 40, 48, 48, 48, 48....

Upsampling to 176.4 kHz

40, 42, 44, 46, 48....

Asynchronous Sample Rate Conversion is like upsampling to any given target.  It does the MOST manipulation of data:

39.5, 42, 44, 47, 49.5 .....

All these techniques were really good before 2000 as DAC's performed so much better with high rez (88.1kHz or higher) data.  In the last few years DACs perform so much better with Redbook (CD, 44.1kHz/16 bit) than the previous generations.