Upsampling put to "THE TEST": R U ready 2 take it?


(WARNING!! Lengthy post ahead! Bail now if you don't have the time or the attention span! ...OK, with that out of the way, let's start...)

I recently got an usampling MSB Gold Link with a P1000 Power Base here on the 'Gon, and auditioned it against my old-school Theta DSPro Basic IIIa that does regular oversampling. I did this not because I was unhappy with the sound of my Basic, but because I decided I had to hear for myself what all the uproar about upsampling was for. The two DACs had very different sounds, especially with upsampling engaged on the Link, so to determine which one was more faithful to the data encoded on the disk, I devised a test that many of you, if you have the right equipment, may be able to try at home for yourselves - if you are interested in knowing what your DAC is really doing to the digital signal it receives.

The first thing I should say is that, after the dust settled, I decided to keep my Basic, because to me it won both the controlled test I'm going to describe, and I prefer the way it sounds. However, I'm not trying to make this post primarily into a review of the two DACs in question here. But I do want to state up front that I don't believe that it would be fair to make the Link somehow representative of all DACs on the market with upsampling. So even though I found that this feature did not improve DAC fidelity in my case, I'm hoping others will be able to try to repeat my test, maybe with other kinds of processors, and post back here with their own reports on this issue.

THE TEST:

Let me explain what I tried to accomplish in a nutshell. I was listening to two DACs, with everything else in the system kept the same, that each gave very different presentations of the music on the disks. For some recordings, I thought one machine might sound better, but for different CDs, I would give the nod to the other. I became dissatisfied with the subjective nature of this alternating preferrence business, and thought up a way to more objectively assess what was going on. This is an audio test, not some kind of "looking the bits" thing, which I wouldn't know how or be able to do. Besides, that sort of test won't tell you anything about the decoded analog sound your processor puts out. I'm sure I'm not the first person to ever employ this test set-up, but I haven't read about anyone doing this before either.

Here's what you'll need to try this: 1) A high-quality analog source, preferrably vinyl; 2) Some kind of analog-to-digital converter (ADC) that outputs CD standard 16/44 digital, preferrably with an analog input level control, like those found in CD-R recorders; 3) Your DAC to test must have a digital input, which will rule out those found in many one-box players; 4) A control preamp with a tape loop and tape monitor function, preferrably with a full-function remote control that allows remote selection of the tape monitor switch.

Here's how to set up the test:

Take the output from your phonostage into the phono inputs on the preamp, or use the preamp's built-in phonostage if that applies.

Using the preamp's tape outputs, send the preamplified phono signal to the analog inputs of your ADC, such as the CD-R recorder I used.

On the recorder, select the analog input monitor function, and adjust the record level control referring to the level meters so that the signal level is in the proper range - not too quiet, not overloading the input.

With the ADC now encoding the analog signal from your phono into 16/44 digital, take the digital output from the recorder and send it to the digital input of your DAC that's under test.

With the DAC now decoding the digitized phono signal back to analog again, send the analog outputs from the DAC back to the tape monitor inputs of your preamp.

With a record playing, listen to the volume when the phono input is selected normally on your preamp, making any room-level adjustments needed via the preamp's volume control, and then engage the preamp's tape monitor button to switch to hearing the decoded signal coming from your DAC, in order to compare the two volumes.

Using the analog input level control on your recorder to fine tune the level attenuation coming into your ADC (and therefore out to your DAC), go back and forth between listening to the direct phono sound and the DAC sound by switching the preamp's tape monitor button, making the level adjustment at the recorder until the volumes coming through your speakers from the two sources match as closely as possible. (If you have a sound pressure level meter and an appropriate test record, you could use this to perform a final level-match check.)

With a revealing test record playing, retire to your listening chair, remote in hand, and you can now make instantaneous level-matched switches between hearing the straight feed from your phono section, and hearing the same feed as encoded and decoded in CD standard 16/44 digital by your ADC and DAC, simply by engaging the tape monitor function on the remote (leave the preamp's input selected to phono throughout).

LISTENING

In this test, the sound of the DAC will be contributed to by the sound of the ADC. The ADC process can be generally more transparent than the DAC process if all other things are equal (they never are) since jitter is not a factor, but a good ADC should obviously be used if at all possible. Any CD you play in your system was made in a similar way, by using an ADC to convert an analog music performance into the code on the disk. In this test, the LP stands in for the original performance - the record actually is our "absolute sound". This means that we aren't primarily concerned here with the usual question of how well the record captures a believable account of a real performance. If it's a revealing record, of course, it probably will have been recorded with a good degree of fidelity, but in this test we're not trying to decide how much what we hear sounds like what we believe it should in real life. Instead, we are only trying to ascertain to what degree the reconstituted version of the record coming from the DAC actually sounds like that record, in a no-delay level-matched comparision. I frankly think that the one-step, real-time ADC conversion process used here has got to be at least as transparent to the analog sound of the playing record as the studio-mic-feed-to-finished-CD process that occurs with the music that's on a disk you buy from the store.

This test is pretty good for listening to the sound of your DAC vs. the phono feed as a way of assessing the accuracy of the DAC, but it is even better for comparing the sounds of two DACs, or of comparing the sound of an upsampling-switchable DAC with it's upsampling turned on vs. turned off. This is because, in addition to the sound of the ADC making its contribution to any sonic deviation from the straight phono feed, there are also two extra runs of interconnect to account for: the run from the preamp's tape outs to the ADC, and the run from the DAC's analog outs back to the preamp's tape ins. Even if you had a perfectly transparent ADC and DAC combo, the extra wire runs would mean that the sound from the loop out would never be the same as from the phono itself. So expect some degradation just from this factor, and use the best cables you can for the test. (This reality is ameliorated somewhat by the fact that one of these cable runs, from the DAC to the preamp, will always be there in normal use anyway - you couldn't listen to the DAC without it, so you might as well just consider it to be part of the DAC's sound for practical purposes. The same is true of your digital interconnect, and any outboard jitter-reduction boxes you regularly employ; use them in this test if you use them for normal listening.) The reason, of course, that the test is better for comparing at least two conditions of digital-to-analog conversion rather than one, is that these other unavoidable factors (the ADC and cables) will always be held as a constant, allowing you to focus in on just the isolated differences between the two (or more) DAC variables under test. With only one, non-adjustable DAC to test, you'll never really know which deviations you hear are attributable to the ADC or the wires, and which to the DAC itself.

MY RESULTS

This was the system I auditioned with:

Modified Technics SL-1200 turntable
Benz-Micro Glider M2 cartridge
Camelot Technologies Lancelot phonostage

Innersound remote preamp

HHB BurnIt CDR-830 CD-R recorder

Monarchy Audio DIP 24/96 jitter-reduction box

Conrad-Johnson MV-55 power amp

Thiel CS2.2 speakers

All Cardas Cross analog interconnects (single-ended RCA)
All Cardas Lightning digital interconnects (single-ended RCA)
Cardas Cross speaker cables
Synergistic Research Master A/C Coupler on preamp

Audio Power Power Wedge Ultra 116 (balanced, filtered AC for all digital front end components and preamp)

(The digital components were used with their stock power cords, as I don't own multiple identical aftermarket cords.)

All right, let's get to the listening test results. Like I said, this is not supposed to be as much of a review of the MSB Gold Link or Theta DSPro Basic IIIa DACs as it is a report on my test set-up, an assessment of upsampling as implemented in the Link, and an encouragement for some of you all to try this experiment too and report back here. So I'm just going to jump right to my conclusions and skip all the listening impressions, if that's OK with you.

First, just a little background: Both of these units sport dual-differential DAC chips per channel for balanced digital operation (lesser Links do not), with multi-bit PCM architechture, 8X oversampling, and op-amp output buffering. The Theta implements output filtering in the digital domain on DSP chips incorporating proprietary software, whereas the MSB boasts of 'minimal' filtering. The Gold Link comes with an installed upsampling board that can be set to 96KHz or 132.3KHz frequencies internally via a removable jumper, and an exterior switch to turn upsampling off. Build quality seemed roughly comparable, nods going each way. The Basic originally cost about $2,700 - about a grand above the Gold Link with its Power Base - but used today the Theta sells for slightly less than the MSB combo used.

I esentially did two kinds of listening, and reached a few significant conclusions. I'll start with the second kind first. The last comparitive listening I did was with the phono bypass test descibed above. I did this part for a few hours, after I had already spent many more hours auditioning the units conventionally (all auditioning was done over several days). The controlled tests basically confirmed for me a handful of things. First, my Theta was the more accurate DAC of the two, and not by a little. Doing the bypass test with the Basic in the loop proved to be remarkably transparent. I really wouldn't want to have to put money on telling you whether I was listening to the direct phono feed or the DAC if you blindfolded me. What I did hear being lost could've reasonably been attributed to the cabling alone, that's how small it was. This made me glad about my CD-R recorder selection, too. Just a slight thinning-out of tone color, a bit less transient jump, and a small addition of background texture were all that was there to let me know I was listening to the processed feed. This also led me to conclude that a lot of what I've always assumed was just a part of the Theta's sound - a little coolness and dryness despite its virtures - is actually on the CDs I play, because I couldn't hear those things in this test. The sound wasn't any cooler or dryer than the records, just a tiny bit paler.

The second finding from this part of the testing concerns the Gold Link and its upsampling choices. As I implied already above, the Link could not do as good a job of disappearing in the test set-up as the Basic did. What is most interesting to me, however, is that it showed its best with the upsampling defeated, meaning that this sought-after option, which has been the main source of all those subjective raves, is actually taking the unit's performance *further* away from reality. However pleasing this DAC's upsampling effect might be on some CDs to some people, it's going to be very hard to argue that a process which takes the encoded sound of an LP record and decodes it so that it sounds even less like that record than when it's switched off, is somehow going to make all your CDs more 'analog-like', as has been claimed.

With upsampling defeated, it was still a lot easier to hear whether the Link was in the circuit or not than with the Basic, but I don't want to go overboard with this impression. We're still talking about maybe a 6% change in the sound now, but I would classify the Basic as being around 2% - and that's including the contributions (and subtractions) attributable to the ADC and cables. The Link was more intrinsically colored and opaque, and more greatly reduced dynamics, extension, and transparency to detail. In particular, it altered tonal color noticably, in a way similar to a classic transformer-coupled tube amp of yesteryear. It rolled the highs and a little bit of the lows, thickened the upper bass and low mids, and gave the midrange a dollop of yellowy-colored warmth. It also pinched the soundstage from the sides and back, and laterally splayed images somewhat.

Switching in the upsampling definitely added more murkiness to the sonic picture - veils were being lowered! The treble was even more rolled this way, but now the deep bass also was pronouncedly lightweight, despite the increasingly thick upper bass. The midrange coloration mentioned above increased in intensity, while dynamic shadings got glossed over. 96KHz was better than 132KHz in these regards - it became sort of a progression from best showing to worst as I moved from regular 16/44 to 24/96 to 24/132, but I have to caution you not to assume that this means that higher is necessarily not better in this regard, since MSB in the manual admits that 132KHz 'overdrives' their merely 96KHz-capable DAC chips. But as it is implemented in the Link, oversampling in this test took the processed sound a few more percents farther away from the reference with each increase in frequency. I do have to admit here that, despite these objective results, more than a couple of times I found myself greatly enjoying the Link's renditions of different CDs in my preliminary auditioning, and some of those most with 132KHz upsampling engaged. However, even before running the bypass tests, I strongly suspected that some pretty hefty accuracy and detail babies were being thrown out with the digital bathwater, so to speak. It's no wonder that the Link's upsampling seemed to be fairly effective at banishing digital texture artifacts, along with such things as harsh-sounding instruments or sterile-sounding CDs, because it was covering up a lot of stuff - with a shovel and an axe - that I guess some folks would rather not know about. But there is a real price to be paid for that, which should not go without being fully acknowledged.

The other main finding I got out of this exercise also involves the upsampling function, and was first run across during my preliminary auditioning before I got to the controlled testing later on. Again, I want to stress that this complaint is not to be taken as an indictment of upsampling as it is implemented in other products which I have not heard, only the Link. But even in reviews I've read of this product, what I'm reporting now has never come up. After auditioning for many days, I decided I had to pick a favorite between 96KHz and 132KHz upsampling, if for no other reason than you can't easily reselect the other option after the unit is installed, but only when the case is opened up. 132k sounded smoothest and most different from my reference, while 96k seemed to preserve a little more of the sound I was used to. I had mostly been comparing either one or the other upsampling mode against my Basic, but now I wanted to compare the two modes against each other, so I completely removed the top cover from the Link in order to have quick access to the jumper on the upsampling board. Focusing on which upsampling rate gave the best overall results the most often, I began to notice a bothersome trend.

When I changed the upsampling frequency and played a song over again, I found that the harmonic emphasis had shifted. This was going on above and beyond any of the more obvious changes or colorations I've already covered. It was probably harder to notice at first, because upsampling in general rolled-off the highs where the overtones live, but just listening to the two upsampled processing choices without interruption from non-upsampled sound brought the phenomenon to the fore. Especially on material with artificially-enhanced harmonic content, such as chorused electric guitar, it sounded as if each setting of the jumper was selecting a different filter for the overtones on the CD. Once identified by the ear, this was hard to ignore, even on acoustic material. The effect was very dependent on the exact source being processed, but seemed disturbingly 'mathematical' in nature. When upsampling was turned off, the selective emphases disappeared, and the harmonics regained a more neutral perspective. (If you listen carefully for this in a revealing system with sensitve test material, it still won't hit you over the head right away, but I do want to see if anybody can confirm this finding. To those who have a Link with upsampling and want to try this, it's easier to remove the top cover completely out of the way if you first unplug the front panel display.)

CONCLUSION

I know that even if everybody did these comparisions and heard what I heard, they still wouldn't all agree with my conclusions, because a lot of folks don't care about whether or not the sound coming out of their system is 'accurate' or not - they take the emminently practical position that they just want it to be pleasing. After all, we are listening for entertainment. For this type of listener, even if an arbitrary 'correction' to the sound on a CD just reminds them superficially of a type of sound they'd rather hear, that's fine with them. And I'm not here to criticize their choice, though I can't listen happily that way. But when certain prominent writers who shall go unnamed preach from the bully pulpit about gear containing "magic bullets" to cure something they label "CD sound", don't automatically take them as prophets spreading the only true gospel. Try this for yourself if you can, and let us know what you find out.
zaikesman
May I join the cohorts of those who thank Zaikesman for the thorough experiment and thoughtful reporting of results.
The link below will connect you to an article on exactly this topic from hi-fi news and record review. It is informative.....at the very least.......nice experiment btw.

http://www.aslgroup.com/dcs/upandover.htm
Hello again. It has been over a year since I originally posted this article, and in it's time it received over 900 views, the most of any thread I've started on Audiogon. To my mild surprise, I have yet to discover any other accounts about anyone else, say a reviewer, constructing a test similar to the one I describe, but I continue to maintain that the basic procedure I outlined should be an objectively valuable method for assessing DAC performance - one that ought to be copied IMO.

My method enables a feasible (provided you have the necessary ancillary gear listed above), common-sense, in-home way to comparision-test - instantaneously in real-time - the sonics of one DAC against a near-absolute reference of the signal it’s decoding, or even better, to compare the relative performances of two or more DAC’s (or of one DAC with two or more selectable operational settings available, such as filter slopes, upsampling rates, etc.) in a manner which allows a positive determination to be made of which option remains truest to the sound of the reference feed, and to identify whatever deviations from reality are occurring with much more certainty than through conventional subjective auditioning alone. Then an audiophile can pick his or her poison, be it literal accuracy or euphonic massaging, without feeling forced to simply wonder in some degree which of two or more ostensibly good-sounding presentations is more correct and which is less correct, in addition to helping aid in quantifying what audible system shortcomings should be rightly attributed to DAC performance in isolation.

[BTW, I am happy to report that I was able to successfully sell my MSB Gold Link/P1000 on Audiogon for what I had invested in them, even after the appearance of this thread. Today I am still using my long-discontinued Theta, and have not tried any other snazzier, 'new and improved' digital sources in my system during the interim, and have no immediate plans to either, be that to my credit or shame.]

My only disappointment with the response to this thread has been that no one has attempted to duplicate the described test set-up and report their findings back here. As this was one of my primary goals upon writing the article in the first place, it is also my main motivation for posting anew today: there are many newer members on Audiogon since the last thread activity, and I wanted to bring this article back up from the depths of the archive in hopes of freshly stirring the pot (might not work too well, but what the heck).

In that spirit, I also intend to re-perform this test set-up again myself, even though at present I can only do it with my one (non-upsampling) reference DAC and no other competition. I want to do this a second time mostly because there have been several upgrades throughout my system during the meantime that should make the test more revealing, and I am curious whether I will still come to the conclusion that there isn't a hell of a lot of meaningful difference to be heard between my straight analog feed and the same feed that's been run through my reference ADC and DAC conversion and reconversion. (Those system changes will be detailed when I post my listening results.)

So I look forward to playing with the test procedure once again, and especially to any and all additions to this thread from the forum contributors. I know it takes a bit of time and effort to run this procedure (not to mention read through this thread! :-) , and that not everyone has the needed gear or system configuration to do so even if the inclination is there, but I'm hoping that some of you will be able and inspired to to run this test for yourselves, and write about it for all of us. (Anyone interested who has further questions about how to do this test for themselves is encouraged to post them here or email me directly for clarification.) Thanks for reading, Z.
Zaikesman,

I have read your long post very, very quickly but I think that I have the gist of it.

Sometime back (say 1 to 1.5 months ago) some other A'gon members & I had a long discussion on upsampling & oversampling. Much EE stuff was discussed in those 35 or 37 posts. Guess what......that thread has been deleted!!! Somebody started a "24/192.." thread recently & I wanted to point him this upsampling/oversampling thread but I discovered that it was deleted!!! I'm quite pissed!

We cut thru the chase in that thread & discussed what upsampling & oversampling do & why the sound of the music was different in each case. It laid bare the basic fundamentals of both techniques. Anyway, it *appears* that A'gon & the sponsors didn't like this very much!

Anyway, with upsampling I'm convinced that one is listening to the digital reconstruction filter (that follows the upsampling event). Each manuf. has their own proprietary technique for doing this. It seems that very few of these implementations are actually liked by the end-user while most of them just add distortions to the music signal.

It is no wonder that you like an oversampling design. This too requires a filter after the oversampling event but since no new data points are being estimated between data points sampled off the CD, the filter is less in the critical path. Hence the reproduced sound is more true to the recorded signal.

Anyway, looks like you have gone thru some pain to discover this yourself. I'll read your post in details & *might* have some more comments to make.
Bombaywalla: I can't pretend to possess the technical expertise to comment authoritatively on your post (though I'm not at all sure you are on the right track when you opine that "very few of these [upsampling] implementations are actually liked by the end-user", and I don't really follow where you talk about a filter being "less in the critical path").

I will say that I tried my best to stress, in all of my posts above, that my findings should in no way be taken as some sort of overall indictment of all so-called upsampling techniques or implementations. To the contrary, as best I can understand it (hard definitions in this area seem tough to come by), the theory behind upsampling appears to make a lot of sense on its surface.

But getting any takers to try to running my controlled auditioning test and shed some more light on the topic is proving a tall order. When it comes to testing upsampling effects in particular, part of this is predictable: there aren't too many DAC models out there featuring selectable/defeatable switched upsampling, and lots of the newer models incorporating upsampling technology are all-in-one players, not stand-alone boxes with digital inputs.

I know I prominently mentioned upsampling in the title of this thread, and that was deliberate. I thought it would attract the most interest to my article, and investigating upsampling's effects in one of the DAC's I was testing was a big part of what I was trying to accomplish when I first hit upon the described test set-up.

But I think the wider implication for employing my test design lies in simply comparing one or more DAC's against each other indirectly by making serial comparisions against a known constant reference feed. Believe me, I would love it if someone with another DAC model featuring switchable upsampling were to run my test and post their results, but I'm no longer crossing my fingers on that wish.