Upsampling DACS: Take the Pepsi Challenge

I emailed MSB yesterday regarding the Link DAC III's ability to INTERPOLATE AND UPSAMPLE. The response was as follows:

Thomas, We do have this available, we call it our upsampling upgrade. It does smart interpolation and upsamples the signal to 24/96kHz as well as 24/132kHz.

This upgrade is $199.00 and can be installed by the customer at any time.

Thank you for your interest, Scott Rust MSB TECHNOLOGY This information lead me to order a Link DAC III with the upsampling and interpolation chip. When I asked the Scott why does MSB not state anything about Interpolation on their website, he said "Nobody Knows what Interpolation is". Anyway. I have Pioneer DV-414 as the transport and the Link DAC III will complement it. I just thought I would interject my ignorance or 2 cents which ever is worth more!

I thought I'd done the laymen thing already, and as you'll probably see, I may still fail at that yet...........As said above by MSB (posted by perhaps an F-14 pilot?), interpolation occurs when voltage amplitude values are APPROXIMATED during this up-conversion from CD resolution data, to much HIGHER resolution data. It's sort of like a line doubler for video projectors...sort of, but not exactly..................I think of it in this simple way: the conversion of the audio data, in the digital domain, to a higher bit and sampling rate (upsampling), allow the high performance DAC to do its conversion on this LARGE amount of data, thus making full use of the DAC's superior resolution. IT'S NOT A MATTER OF CREATING NEW DETAILS IN THE RECORDING that were never there to begin with, it's a matter of getting the most out of what was always there...just like everything else in this hobby.......................It's also not merely "digital" we're talking about, but rather how the digital data gets turned into the analog voltages/waveforms...before it goes to your amp, or preamp. THAT'S ALWAYS THE MOST IMPORTANT THING IN DIGITAL AUDIO. Otherwise, you're making assumptions like my smug, MIT graduated, aeronautical engineer uncle. He's said the cliche right to my face, "but digital is digital; how could one CD player possibly sound different from another one?" You know, the old/stupid "bits is bits" argument, only he didn't even bother to think of it on that level............No DAC is perfect, but it is the critical "roadblock-weaklink" in the digital playback chain. Therefore, if you can make use of a "superDAC" on "mere" CD audio, it's much better than using DACs that operate only on the level of "CD quality" data................Anyway, audio is always about maximizing your UPSTREAM performance, in order to make full use of what you have DOWNSTREAM. UPSAMPLING "guesses" at many extra millions of possible "loudness" levels (and frequency "pulses" in time) in the digital data, BEFORE IT EVER GETS TO THE DAC (that needs all the "help" it can get)..............As perhaps most who'll read this know, SACD uses a different digital process, that samples at nearly 3 million times a second, and only uses one "loudness" bit, to tell if the waveform is rising or falling. It depends on the sheer density of those 2.8 million pulses every second, to describe how quiet or how loud the music is...............It's all much more complicated than this, but oh well, I'm not the real expert here...either from a designer's viewpoint, or from a journalist's viewpoint.

Carl, that was very helpful.Thank you for taking the time to offer such a clear summary. I'll get back to you all after I hear the Bel Canto DAC1 next week.

The following is the actual e-mail (in entirety) sent to me (this morning) by Resolution Audio in response to my question concerning upsampling / oversampling (the question was almost verbatim to my original question above). Indeed, there is no technical difference between upsampling and oversampling. The only difference I can discern is in the marketing. Indeed, digital filters can be very aggressive above the audio band without the adverse effects that analog brick-wall filters have. This is possible because of FIR (finite-impulse response) filters, which have constant group delay (zero phase effect vs. frequency). There is no physical realization of an FIR filter in analog. Using FIR digital filters allows the analog filter to be relaxed significantly, because the first "images" are located at much higher frequencies. In our cd55, we use a passive third order filter which is down only 0.2 dB at 20 kHz, yet the rejection of the images at 700 kHz is about 60 dB. And indeed, the digital filters do not create information that may have existed before the mic feed was converted to digital. Some external "upsamplers" may by their nature apply some other filter/eq, but this is independent of the a/d - d/a process. You are also correct regarding the delta-sigma dacs. These dacs are rated for maximum input rate, currently as high as 192 kHz. These converters all run at the output at much higher rates -- typically 12 MHz or thereabouts. The better ones from Analog Devices use extra filter stages when the input rate is lower. Essentially, the dacs run, say, 256x at 44.1 or 48 kHz, 128x at 88.2 or 96, and 64x at 176.4 or 192. This puts the noise modulator heart of the converter at the same frequency regardless of input. The best multi-bits, including the PCM1704, run upwards of 800 kHz, which allows 16x at 44.1 (and 8x at 96, and 4x at 192 input rates). In sum, your perception of "market jargon to draw interest" is dead-on. In addition to preying on the consumer base which generally does not have engineering degrees (and some manufacturers as well), these products offer the opportunity to sneak in digital eqs which will absolutely sound different. Better? That's a different story. Finally, we have just started talking to a dealer in Indiana. If all goes well, I'll pass along the info in a couple of days. Regards, Jeff Kalt Resolution Audio resaudio@ix.netcom.com

Carl, it seems that Resolution Audio wasn't frustrated by my question. Why must you?