Upsampling. Truth vs Marketing


Has anyone done a blind AB test of the up sampling capabilities of a player? If so what was the result?

The reason why I ask because all the players and converters that do support up sampling are going to 192 from 44.1. And that is just plane wrong.

This would add huge amount of interpolation errors to the conversion. And should sound like crap, compared.
I understand why MFG don't go the logical 176.4khz, because once again they would have to write more software.

All and all I would like to hear from users who think their player sounds better playing Redbook (44.1) up sampled to 192. I have never come across a sample rate converter chip that does this well sonically and if one exist, then it is truly a silver bullet, then again....44.1 should only be up sample to 88.2 or 176.4 unless you can first go to many GHz and then down sample it 192, even then you will have interpolation errors.
izsakmixer
Sean, in fact, Jeff Kalt of Resolution Audio was marketing and using "upsampling" in his players at the time, so yes I do think some manufacturers will be honest when asked directly.

But more to the point, what do you think is so magical about 96kHz or 192kHz? Why not 88.2 or 176.4 or 352.8? I think the obvious answer is that the high rez format in DVD-A is either 96kHz or 192kHz...marketing anyone??

If you could, would you please contrast your upsampling dot graph with the equivalent oversampling dot graph? Remember that to get to 96kHz from 44.1kHz in your example you have to increase the number of dots from 20 to 43.5 dots. What you described is essentially a 2x oversampling routine with linear interpolation. The graph cannot get any smoother than the original unless you use something other than linear interpolation. Yyou are just connecting a series of dots in a line between samples, otherwise.

The main reason for adding points between the original samples is ultimately allow a more gentle analog filter. The original (really bad souding) CD players used no oversampling and analog brick wall filters to avoid the problems associated with the Nyquist limit for 44.1kHz sampling (22.05kHz) and the spurious images that get reflected back in band. These sounded horrible and led to 2X, 4X, 8X etc oversampling moving these images well beyond the audio band and allowing more gentle (better sounding) analog filters.

To paraphrase Charles Hansen, adding another digital filter (upsampling) to the chain will affect the sound; however it is certainly possible to design a single digital filter with exactly the same composite characteristics as the two cascaded filters, usually justs costs a little more money.

Anyway, not trying to be a pain in the a$$, I just think the marketing component of the choice for 96kHz or 192kHz needs to be pointed out.
The connect-the-dots metaphor is really unfortunate, because a lot of audiophiles buy into the idea that that's what a DAC does. But reconstructing an analog wave is nothing like connecting dots. More dots DOES make it easier to DRAW a wave. But as long as you have enough samples for the bandwidth, a DAC can reconstruct that wave without more information.

(BTW, the example given above didn't have enough information to do so, because it called for only two samples per cycle. You need fractionally more than two to reconstruct the wave properly.)

Imagine that, instead of a wave, you were trying to trace a straight line. The more dots you had, the easier it would be to do this freehand. But a graphing calculator would only need two points.
Pabelson...According to Nyquist, just two (error free) samples per cycle will perfectly recover a sine wave. But, in this error-prone nonsinusoidal world, where I played with digital data stream representations of analog waveforms (non audio), experience taught me that four samples per cycle was worth the trouble. That's why the 96 KHz PCM of DVDA (or 192KHz for stereo) solves the bigest problem with redbook CDs. 24 Bits is nice too.
You guys are all correct. Where i "fell down" on this one was that i was thinking in the analogue realm rather than in the digital realm. When i was thinking of how to explain "upsampling", i was trying to demonstrate exactly how "non symmetrical" a musical waveform really is. This is why i suggested that the second set of 20 dots / samples be placed randomly rather than in a neat and orderly fashion between the other dots / sample spaces.

What i forgot to take into account was that we weren't dealing with analogue here at all. We are dealing with analogue that has been hacked to bits ( literally ), completely butchered as it was converted into another format and is now trying to be re-assembled as best possible back to what it was originally. Kind of like taking a fish, throwing it into a blender and hoping to re-build the fish once it comes out of the blender. Good luck.

If you doubt this, try looking at some of the waveforms that Stereophile tries to reproduce on various digital devices. Given that these are symmetrical test tones, you can only imagine how poorly some of designs / devices would do with more dynamically complex musical signals fed into them.

As such, there is little resemblance to what the original analogue waveform looks like after digital processing due to a LOT of various factors, some of which have been more than amply pointed out above. I apologize for the mistake and would like to say "Thank You" to those that corrected my mistakes.

Having said that, i'm glad that at least part of what i was trying to convey was understood and not completely lost. To be specific, i'm talking about the various types of filtering and cut-off frequencies used, why this area of operation affects what we hear inside the audible bandwidth, etc...

As to Germanboxer's comments, i agree that the majority of upsampling is based on parts that are already commercially available products. This not only makes things easier to design, it is also cheaper to produce. Otherwise, manufacturers would have to build "one off" devices for each product manufactured, which would make every upsampling DAC a custom built piece. While this would probably result in better quality as everything would be designed from the ground up rather than just using what was already available, it would also be horrendously expensive to produce, especially in very small quantities. By relying on parts / circuitry that is already in production, at least a portion of the benefits of such an approach can be had and prices kept within the "working man's" budget. Even then, some "working men" may still have a problem with the price on some of these units. Sean
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Not quite, Eldartford. A digital system cannot accurately reconstruct a wave that is exactly half the sampling frequency. That's why I said the sampling rate had to be fractionally higher (granted, a very tiny fraction) than twice the highest frequency. In the example given, there were exactly two samples per cycle, and that wouldn't work.

And just what is it you think is "the biggest problem with redbook CDs"?