MUSINGS: On Experts, Experience, and Opinions...

So last night, instead of going to bed early as I was supposed to, I decided to have a look at this interview of Allen Sides from Ocean Way Recording on TWiT.TV. As usual, Scott Wilkinson does a fantastic job with the interview and takes questions from the audience.

Obviously, Mr. Sides is a man of many years of experience and can speak authoritatively on MANY topics related to audio hardware, studio production, and historical anecdotes based on those years.

But some things bothered me. Around 16:00 there was talk about DSD: "somehow between making the recording and that SACD it doesn't sound quite as good as it should". Really? By 17:00, there's discussion about CD copies, different stampers sounding different, then an anecdote on Mariah Carey and how the pressed CD sounded bad compared to the reference. Okay... Maybe... How about someone ripping the disks and comparing the data integrity and talking about that? Given that Mariah was married to Tommy Mottola until 1998, this anecdote is now at least 16 years old so any hope of forensic assessment is long gone - does it still apply as a generalization these days?

Things get really bizarre by 26:00 - "I have never been able to even make a copy of a CD that sounds as good as the CD I started with... It always sounds worse". As you can see, Mr. Wilkinson was perplexed and commented that "it's almost like generational loss in analogue" that Mr. Sides is referring to. "Multiple degenerations"? Please...

I wished Mr. Wilkinson would have done a follow-up question like - "What if you bit-perfectly ripped that original CD to a computer - does it sound the same then?" "How about if you copy to a different hard drive, does it sound different?"

It's also clear that there are some limits to Mr. Sides' knowledge/experience which most of us in the hobby world would have no difficulty discussing. (34:30) Q: "What's your idea of FLAC encoding?" A: "I'm really not that familiar with it." Fair enough, a person cannot know everything.

Although I'm not that old at this point in my life, I have learned some things in my "travels" both personally and professionally. One which I hold dear is that no matter how much we can respect and trust the "experts" for their lived experience and knowledge, they (like us) are all just human. And as humans we all have idiosyncrasies and biases. In this case, I don't think it's much of a stretch for any of us who have spent hours on our audio systems, used EAC or dBpoweramp for ripping to ensure bit-perfect copies, to stand up with good confidence and tell Mr. Sides that he's just plain wrong about not being able to make a copy of a CD that sounds identical. The fact is that in more than 30 years of the existence of the CD format, there has been no evidence of this when variables are controlled for (eg. ensuring that bit perfect copies were achieved, the rater was blinded, etc.). If indeed "generational losses" were possible with digital, this would already be a well known fact and there'd be no uncertainty whatsoever! Moreover, if this were fact, it would change significantly how we deal with accuracy of our digital data (hey... how can I be sure that those numbers in my bank account are accurate?!). Experts can provide educated opinions, but ultimately they are just opinions and not necessarily fact. The same goes for his strange comment about SACD not sounding as good as it should between studio and the physical disk. What is more likely, that the digital data somehow mysteriously changed over time or that his own psychological expectations changed as the memory of the live studio event consolidated? (Assuming of course that the data wasn't altered by some mastering engineer along the way.)

In this world, there are many mysteries yet to be discovered and likely much we as a species will never know. But digital audio systems which are inventions of the human mind based on mathematical constructs and technologically engineered devices (like the CD) that ultimately changes the physical world (sound waves) were not produced by serendipity. I do not feel it's good enough that we should just shrug our shoulders and declare some experiences to have enduring truth like parts of this interview. That surrender to logic leads us into the realm of "anything is possible!" and ultimately the slippery slope on the path to "snake oil". This is especially significant in the impact on those already obsessive-compulsive and perfectionistic (ahem, like many audiophiles). Maintaining an objective approach hopefully allows a counterbalance to this. An opportunity to take a step back and question the things which seem to make no sense. An opportunity to explore reality with techniques and at times instruments of greater sensitivity than that which we are endowed with within this mortal shell. Although the human mind is the best "instrument" to perceive the beauty of music, accuracy of the reproduction chain is a different matter and can be detected by the use of objective techniques with obviously greater sensitivity.

It's fun listening to interviews like this and I certainly felt it was time well spent nonetheless.

MEASUREMENTS: Another Look - Audioengine D3 clipping at 100% volume.

The ability to interact with you guys over the last year or so running this blog has been vastly educational for me! Whether data gathering with the MP3 test last year or comments and suggestions with each post... In general I do try to keep up with comments if I can but after 80+ posts now, I apologize if there are some comments I've missed along the way.

The post today is thanks to the keen eye of "Solderdude Frans" and his comments & suggestions to the previous post on the Audioengine D3. As per Addendum 2 in that post, indeed there is indication that a 0dBFS signal is clipping with the D3. I had missed it due to the fact that I was looking at the square waveform without considering the possibility of clipping contributing to the shape observed. Also while calibrating for the RightMark tests, RightMark and E-MU were flashing red for clipping and I thought it was that the amplitude was too high for the E-MU 0404USB rather than considering the possibility that the clipping was actually from the D3 output itself.

So, as suggested by Frans, this is what a 0dBFS 1kHz sine wave looks like with no volume attenuation (ie. 100%):

Yup, the peaks are being clipped!

The point where the clipping ends is with the hardware volume control turned down to 92% (93% almost was good enough) in the Windows 8.1 control panel (software volume level in foobar stays at 100%):

I re-ran the RightMark measurements ignoring the warning about clipping to see what happens to the measurement results:

As expected, there's a deterioration to the measured THD and IMD results going from 92% to 100% with clipping - still low using the RightMark methodology but more than 10-fold worsening with the clipping in place.

Conclusion:
Interesting... Like the first sample AudioQuest Dragonfly reported in Stereophile awhile back, it looks like one needs to back off the Audioengine D3's volume setting a bit in order to avoid clipping. In the case of my sample, pulling down the Windows volume control to 92% did the trick.

Well, this fact obviously blemishes my impression of this little USB DAC as an accurate audio device. In practice, it's unlikely I will push the volumes to 100% but I wonder if this was by design. I find it hard to believe that the engineers did not check the clipping characteristics. Rather, my suspicion is that the engineers felt it was OK to allow the signal to clip a bit to increase the perceived amplification level. In a noisy environment, the extra amount of amplification of low level signal at the expense of distortion might be a reasonable trade-off that may not be too noticeable (possibly no problem at all if the music is soft and rarely hits 0dBFS) - of course, this would not be a "high fidelity" practice.

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Seems like there are some ruffled feathers around the recent "Geek Out vs. others" measurements published by Light Harmonic. Good to see that the Geek Out is being designed with good measurements in mind; encouraging. I can see how other manufacturers can be a bit unhappy about all this... Although it's probably wise to have a 3rd party perform the tests rather than a direct LH release, I suppose if it gets the manufacturers thinking about better engineering and competition in this regard, that's a good thing. Curious that there was no frequency response measurement. I wonder if at 44kHz, the Geek Out does measure like a NOS DAC with some early roll-off in the high frequencies.

Ho! Larry somehow believes that "It's proven wrong to assume people could hear only below 20KHz" (see the comment section of that article). Ok. I suppose Ashihara's paper from 2006 could be used to argue this. But we're talking SPLs around 80 dB at 20kHz for most subjects as the threshold of hearing pure tones and there's quite a significant jump between 18 to 20kHz. Threshold for hearing pure tones isn't exactly evidence that this is beneficial for real music! Furthermore, if you look at the test subjects, they're aged 18 to 33 and the majority of these are young women! Good that hi-res can satisfy the golden-eared audiophile lady in our midst... Unfortunately it's not going to do much for the old boys I suspect ;-).

MEASUREMENTS: Audioengine D3 USB DAC / Headphone Amp Reviewed.

In the last couple of years, we have seen a proliferation of small sized USB DACs. Devices small enough for laptop-sized portability aimed at the headphone user who wants a bit more power to drive better 'cans' and provide improved sonics than what's available through the laptop's phono jack.

I guess it must have begun with the AudioQuest Dragonfly back in 2012. Currently in the 2nd incarnation as version 1.2. Version 1.0 got quite a positive review in Stereophile among other magazines which has helped propel this class of device into audiophile acceptability (if not some mainstream approval?). With the Kickstarter success of the Geek (Out) last year, these devices continued to gather press.

To create a device like this can be difficult given the dependence on the USB port for power. I've certainly experienced first hand the noise pollution from plugging my TEAC DAC into the computer USB port and this being picked up by my Emotiva XSP-1 preamp's analogue passthrough. That was why I bought the USB-to-ethernet cable extender to provide some noise isolation that thankfully worked.

So, a few weeks ago as I was perusing the local computer store, I ran into a sale on the Audioengine D3 USB DAC. I figure, what the heck - at less than $180CAD, it'll give me something to play with and if it sounds good, maybe it'll accompany me on overseas trips with a good pair of headphones.

MUSINGS: "The CD can be said to be an analog disc, just like the LP" (UHF Magazine, 2014)

(On occasion, I will refer to copyright works under the principle of "fair use" for the purpose of commentary. This is one of those posts.)

I don't get a chance to hang out at the local large bookstore chain (Chapters) very often. When I do, I will wander over to the magazine rack and see what's on offer for the month. Here in Canada, there is one (and I think the only) magazine catered to audiophiles published in the country - UHF Magazine (Ultra High Fidelity) from Montreal. Here is Issue #94 (March? 2014):

So it was with interest to see an article on "A critical look at the Compact Disc for music delivery" referred to on the front cover written by Mr. Paul Bergman titled "Inside The Compact Disc". I figure it might be interesting to scan the article quickly to see if there's anything new to be gleaned... Alas, within a couple of minutes of scanning page 25, this paragraph just jumped out at me from the article:

To save space, individual binary digits are not stored. The length of a pit corresponds to the number of "one" samples before a "zero" is encountered. The length of a pit or land is of course an analog value, and for that reason the CD can be said to be an analog disc, just like an LP.

Wow... Ponderous man... So a CD is a material thing. All material properties have variability - the length of a "land", the depth of a "pit", the transition between "lands" and "pits". So what? As long as it's within the margin of error within specification, a one is a one and a zero is a zero. No problem. It's not like the length of a pit is going to end up with any other interpretation than either 1 or 0 by the decoding circuitry. It's binary; there is no 0.5. It's digital. The author even talks about the use of Reed-Solomon coding for error correction and mentions interpolation when errors are detected.

Here's another nugget from the same article (p. 26):

Though the CD is not, as I have explained, a purely digital medium, some true digital media do exist. Reference Recordings, for instance, uses a data DVD to distribute its 24-bit 176.4kHz music files...

So data DVDs are "pure digital" and (audio) CDs are not? Does the author not know that "lands" and "pits" also underlie the physical structure of a DVD as well (higher track density of course)?

I assume that the author would consider CD-ROM "pure digital" since it too contains data (as if digital audio is not data) and DVD evolved from it?

Presumably the author doesn't understand that the only difference between the "Red Book" (1980) CD format and CD-ROM "Yellow Book" (1988) is in the data structure. The older Red Book format is much simpler - an inner lead-in segment which contains the TOC (Table of Contents) that gets read when you first insert the disk. When you select the track, the player will look for the time code corresponding to the start of the track and start playing music through to the end of the track sequentially. (The data stored in the TOC is contained in the ".cue" sheets in CD rips.)

Yellow Book and all that followed, leading to the DVD-Books and modern Blu-Ray Disc formats have a much more complicated structure which contains an actual file system; typically ISO9660 or UDF. This adds the extra layer of complexity and features like file names, permissions, directories, opportunity to append/delete/move for rewritable media, etc... Error correction is also much better at the expense of some storage space. I can only assume the author of that article somehow associates these characteristics with "pure digital".

How strange... In 2014, a "technology" magazine would even declare that the Audio CD is somehow not a "pure" or "true" digital medium. I guess the editors must have missed this! Hopefully the readership can send a few "Letters to the Editor" to set this straight.

Well, it's only been about 32 years since the reflective polycarbonate disks came out...

Addendum:
Perhaps the author is taking to heart the title "The Analogue Compact Disc" from Robert Harley writing for Stereophile a little too strongly! That article was from 1994 and speculated about jitter rather than questioning the digital-ness of an audio CD. Even in that article, I'm concerned about this declaration: "It's becoming incontrovertible that CDs containing the same 1s and 0s produce varying levels of sound quality." Incontrovertible. Really?

Recalled Article on Vinyl Rips and High DR...

Hey guys, I "recalled" an article I posted this AM on why DR increases with vinyl rips based on RIAA EQ processing.

The reason being I think small changes like +2-4dB of DR in compressed recordings using the RIAA EQ are very much based on the resolution of the emphasis and de-emphasis filters. JR_Audio made a comment on this and I had a look at the accuracy of the RIAA curves I made. Indeed, mathematically essentially perfect curves <+/-0.1dB add very little DR extension, maybe only 1dB whereas "looser" accuracy like +/-0.5dB through the audio band could give +2dB with the DR Meter with some idiosyncrasies depending on the song and which frequencies the RIAA inaccuracies lie.

The problem is that these are simulations done on a computer with accuracy of 32-bits. As I said in the original article:

In real life, things are more complicated. Even without a significantly better source master, for the final vinyl mix, small changes can be done to the signal such as mixing all the bass frequencies up to 100Hz into mono or taming of the high notes along with the RIAA curve. The results might be more pleasing or euphonic (see this article on mixing for vinyl and how much work potentially needs to be done). On playback, other physical factors are also involved such as the tracking force of the cartridge/tonearm or idiosyncrasies of the cartridge and accuracy of the phono pre-amp. Also, post-processing such as noise/click/pop reduction would add another variable for the results from "needle drops". If you look at a number of different vinyl rips, it's quite common to see slight DR variation depending on the equipment and technique used (typically another DR +/-1dB is not uncommon).

Indeed, I noticed this afternoon playing with Audition just how easy it was to artificially inflate DR values! For example, mixing all low frequencies <100Hz as mono. I played around with this using Lorde's Royals song using a lowpass filter to isolate those low frequencies and then mix them back into the highpassed upper portions as mono... Even though the sound wasn't significantly different when volume matched, it was not difficult at all to get a DR8 original 24/48 HDTracks version up to DR13. As I noted in the original article, I believe this is just the result of removing peak limits and clipped portions, allowing the calculations to extend these portions with the DSP operations.

Given the effect I saw, ultimately I think there's no conclusion one can draw between what's measured on a vinyl rip and knowledge/proof of whether truly new masters are being used unless specifically told about it!

I guess after trying this for awhile, I realized just how sensitive the DR Meter can be. Ultimately it is useful as a tool to explore the average dynamic range of albums and especially pick out very poor masters with strong peak limiting. It's also useful to determine if 2 pressings are exactly the same.

Anyhow... I might review this later if anything clears up for me in the days ahead; just really hard to post something unless good conclusions can be reached.

I'll leave you with the last bit of the post however...

Parting example:
One example of a vinyl remaster that's truly the "definitive edition" is Stadium Arcadium by the Red Hot Chili Peppers. As usual for Vlado Meller the CD's mastering engineer, the dynamic range compressor was stuck at "11" resulting in a sick album with an average DR5. The vinyl release had proper mastering treatment by Steve Hoffman resulting in a DR12 average. That's the kind of difference one wants to see between CD and vinyl to be sure of truly better vinyl source material.

Have a good weekend everyone. Now go enjoy the music and the weekend... Careful about the Pono droppings this week. :-)

Addendum:
Speaking of Pono, check out all the new "limited edition" Pono players you can get on Kickstarter today with one's favourite band named on the unit! Wow, looks like they're going all out on moving as many virtual units as possible sight-unseen except for a few pictures and ear-unheard... It'll be interesting to see what the final product reception will be like when shipped. (I'm also very curious about the objective measurements!)

MEASUREMENTS: Does "Burn-In" / "Break-In" Happen for Audio DACs?

Jazz at Lincoln Center Orchestra with Wynton Marsalis did a great job last Saturday night at the Chan Center here in Vancouver. A friend once told me "Jazz should be seen as much as heard!" I think he's right. Always great to watch artistry in the making and correlate the sounds heard with how it was done. It's also a good opportunity to check out the acoustics in a moderate sized venue with minimal amplification of a 15-piece jazz band. Puts into perspective the dynamics and detail of what one hears in the home system.

Some writers seem to idealize live performances, but IMO, more often than not, what I hear on the home system is clearly better - cleaner, more defined, often much more enjoyable assuming the recording was a good one. The best performance and best seat of the house every night! Of course a studio record could never (and is not supposed to) capture the live ambiance or variation that a live performance can provide. That too is often a good thing as I reminisce on some concerts I've been to sitting beside rather annoying concert-goers. :-)

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I realized something the other night after measuring the Belkin PureAV PF60. I have been measuring my TEAC UD-501 DAC numerous times throughout its lifetime with me. I bought it back in early May 2013 and it has been in use for measurements, headphone listening in the evenings, and as part of my media room since then. I've run all manners of signals through it from standard PCM to high-resolution PCM to DSD64/128. It has been on for days playing "background" tunes as well as much more attentive "serious" listening through numerous albums of genres from jazz, to pop, to blues, to hard rock, to classical...

As of this writing in early March, I've had this DAC for 10+ months. I'll very conservatively estimate that I've put on >300 hours of actual audio through it (not just time turned on). I made sure when I first bought it that I would measure it within the first couple hours of use so that one day (now), I can go back and do a comparison to see if any kind of significant "break-in" can be demonstrated.

Note that the test set-up isn't exactly the same... Components are different like the playback computer, the RCA cable, and I'm in a different house as well. But the setup is comparable:

Win 8 PC --> shielded USB --> TEAC UD-501 --> shielded RCA --> E-MU 0404USB --> Shielded USB --> Win7/8 measurement PC

Results:

So, without further ado, here it is at 3 time points - within 2 hours of use, around 200 hours last year when I moved home, and just about 2 weeks ago with about 300 hours of use...

Summary

Frequency Response

Noise Level

THD

IMD

I figure there's no point measuring at lower resolution than 24/96 for something like this. You will note that the stereo crosstalk has a 4dB spread from highest to lowest (remember, we're talking down at -90dB here). The reason is simple. These are different RCA cables. At "<2 hours" and "~200 hours", I was using a 3' length of RCA cable whereas the ">300 hours" measurement was done with a 6' cable due to the inconvenience of a short cable in the current set-up. As I showed in the RCA analogue interconnect test, length of cable makes a significant difference with stereo crosstalk measurements (shorter is better) for the standard zip-cord type I'm using. This could also be the reason for the slightly lower noise floor especially notable on the THD and IMD graphs above (again, remember we're looking at the -120dB level here!). Otherwise, I see no evidence here of a significant change that would be audible.

Conclusion:

These measurements suggest that there really is no such thing as audible "break-in" for purely electronic devices like DACs within a reasonable period of time. People hearing the "effect" on a regular basis are more likely to be changing their psychological expectations over time than the device actually changing sonic character. I guess change could be happening within the first 2 hours but one almost never hear people claim this; for the most part, people recommend something like 100+ hours. Logically, if anything, electronic devices deteriorate in time as components (like capacitors) get old and connectors oxidize.

Sonic change for mechanical devices like speakers would make much more sense... InnerFidelity had an article about change in the sound of the AKG Q701 headphones over time (65 hours). The measured differences in those graphs are much more than what I'm demonstrating here.

In summary... I wouldn't be worried about "burn-in" with purely electronic devices. If you like the sound at the start, great. If not, maybe give it some time for your ears/brain to adjust and see if you like it then. Sure, keep the device on, blast some hard rock or play a burn-in CD if you feel this helps.

If there's no difference with complex electronic devices like the DAC, it'd be quite unreasonable to expect to hear a difference with totally passive "components" (eg. wire/cable burn-in). I find it suspicious that some companies like this one would claim cables needing 400-500 hours (17+ days straight!) to break-in! The more cynical side of me wonders if there is a benefit for companies to do this because it gives them a "grace period" to tell customers to wait. Furthermore the message itself promotes expectation bias towards improvement in time... "No worries! Give it some time to really sound it's best, Mr. Audiophile!"

Subjectively, I cannot say I've ever thought I could hear burn-in. The TEAC sounded good to me from the start and I'd be foolish to claim with certainty any difference at this point almost a year down the road unless of course there were some kind of night-and-day change (which there obviously hasn't been).

An observation - why is it that "burn-in" essentially always results in a reportedly better / smoother / less harsh sound? How does the component "know" that it should go in the right direction? It's not like the electronic component is functioning like the cells of the body where there's a homeostatic mechanism directing the 'healing' towards optimal functioning... Unless of course, we are dealing with a biological mechanism - the ears & brain. ;-)

Perhaps the standard measurements I present here are unable to capture whatever change there's supposed to be. As usual, I propose to those who are certain that burn-in happens to present any links to information or data to support this belief.

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Listening tonight:
Sonny Rollins Way Out West - just got reacquainted with this old 1957 jazz recording. A fantastic vintage recording from the golden age of analogue done with the tube Ampex 350 tape recorder. This was chosen as the first CD release by Mobile Fidelity back in the mid-1980's (I think 1984). There have been many reissues of this over the years and I think the highest resolution one would be the Analogue Productions SACD from 2002.

Enjoy the music everyone...

FOLLOW-UP: Anomalies in Beck's "Morning Phase" (HDTracks 24/96).

After I posted the last note on "High Resolution Audio Expectations", I got a few "tips" to have a look at some tracks on Morning Phase in more detail... So I got my friend who informed me about the DR6 from HDTracks to send me some pictures:

Track 4 - "Say Goodbye"
Spectral frequency display:

Looks like there's almost nothing after 22kHz but low level noise. Essentially a 44kHz "upsample" for many of the notes. Actually, my friend says a number of tracks he looked at is like this; for example tracks 3 and 5 also (I don't think he checked every track). Since this is a multi-tracked recording with synthesizers and various studio effects, this is actually not surprising - many synth/pop/rock albums are like this. Many samplers and DSPs operate in the 44/48kHz domain so what's laid down is "limited" and there's just no 'genuine' 96kHz sound available.

Track 10 - "Phase"
Now this is interesting:

Track 11 - "Turn Away"
And this:

My word... It looks like tracks 10 & 11 are sourced from some kind of lossy original! Notice the characteristic low-pass from about 16kHz! Might as well be the output from LAME 320kbps (on second thought, 320kbps usually retains up to a full 20kHz with the psychoacoustic model so we're likely looking at 192-256kbps).

Now I'm definitely going to be purchasing the CD rather than any high-resolution download (assuming I want it... Haven't listened to any samples yet). I don't know if folks have checked the CD or if the Qobuz version is any better.

Seriously, HDTracks... Do you guys ever look at these files for quality control purposes before declaring the album fit for high-resolution and charging folks $18USD? I know you claim to just sell what the label gives you, but isn't it a bit disingenuous to be calling much of this album "Audiophile 96kHz/24bit"?

ADDENDUM (2014-03-05):
Given the publicity Beck's album received a week ago and now the revelation of the quality issue, I do hope this serves as a meaningful wake-up call to HDTracks specifically and the audiophile download community as a whole.

If HDTracks sticks its head in the sand on this, it would be telling that their commitment is clearly not to their customers and they certainly cannot maintain any quality control to assure their product is "HD" in any sense of the acronym. Upsampled 44/48kHz is bad enough, but lossy sold as "HD" is just ludicrous!

Thankfully, I did not buy this album from HDTracks... I suppose if I did, I'd be asking for a refund ASAP or at least demanding a higher quality download.

MUSINGS: High-Resolution Audio (HRA) Expectations (A Critical Review)...

I guess an LP can still be considered "high fidelity" by some in 2014... But doubtful it should ever be called "high-resolution"!

Like DSD / SACD, high-resolution PCM (24-bit, 88kHz+) in the form of DVD-V (up to 24/96) and DVD-A has been widely available for more than 10 years already ("rebadged" recently as HRA for "High Resolution Audio"). DVD-V's with 24/96 audio tracks could be easily ripped back in the early 2000's, and by early 2007, the DVD-A copy protection was overcome allowing easy DVD-A ripping and evaluation of the sonic data up to 24/192 2.0 and 24/96 5.1.

In 2010, out of curiosity, I ran a little foobar ABX trial using the equipment I had back then to see if I could tell the difference between 24/96 and 16/44 and posted this on Audio Asylum. The conclusion... I didn't think I could. (I've included that post below as Appendix A for completeness.)

MEASUREMENTS: Belkin PureAV PF60 Power Conditioner.

Happy Friday everyone.

After the negative results for the measurements of the Synergistic power cables earlier this month, I thought I'd just end the month with more power 'tweaks' and do some measurements of my Belkin PureAV PF60 "Home Theater Power Console".

Currently 119.6V, 2.7A being drawn.

Stock photo showing the rear. (BTW, you can't tell from this photo, but the power cable is thick as my garden hose!)

I think I paid about $250CAD for this back in 2007 or thereabouts so it's not new at this point. It certainly has served me well as a 13-outlet fancy powerbar with surge protection. You can check out Belkin's webpage on this item and get the mumbo-jumbo sales talk. I have no idea what a "Level 4" power protection is or what a "Phase 6 PureFilter" does or in what way the "HiCurrent" outlets differ from the others. I do as it says and plug my TEAC UD-501 and Transporter into the "Digital Filter" bank, the 2 monoblock amps into the "HiCurrent" ones, etc. The extra outlet on the front has been convenient as well to plug in the occasional charger or for the laptop. The main display up front tells me the voltage and current being drawn. From my listening position I can't really read the display well so I usually keep the blue LED dim so as not to distract.

The test is simple - how does the TEAC UD-501 measure either plugged into the PureAV PF60 vs. plugged into a reasonable surge protected powerbar (APC P7V ~$25 in this case)?

Standard setup:

HTPC --> shielded USB --> TEAC UD-501 (plugged either into PF60 or APC powerbar) --> shielded RCA --> E-MU 0404USB --> shielded USB --> Win7 laptop

To create a bit of noise in the power system, I turned on the 2 Emotiva XPA-1L monoblocks, Emotiva XSP-1 preamp, LG 55" 3D HDTV, Onkyo TX-NR1009 receiver, Transporter, Behringer DEQ2496, gigabit router, all the pot lights in my sound room on. The amps, preamp, AV receiver, Transporter were plugged into the Belkin PF60. The TV, gigabit router, Behringer DEQ2496 were plugged into the APC P7V powerbar.

The audio system was playing some Simon & Garfunkel at moderate volume through the Transporter during the testing.

Results (24/96 only):

Summary

Frequency Response

Noise Level

THD

Crosstalk

Summary:

Well, nothing much to see here folks... Really minimal differences between whether I plugged the TEAC DAC into the "fancy" Belkin PureAV or the inexpensive surge-protected powerbar.

I did the test around 10:00PM so maybe the power line at this time of the night wasn't all that noisy here in Vancouver. The Belkin PF60's squiggles actually show a little more noise than the inexpensive APC powerbar... Likely just small inter-test variability and I didn't bother measuring a few times since the difference was so minor. It's also possible that the 2 monoblock amps and the Onkyo receiver plugged into the PF60 were noisier. However, I would have thought the 55" TV plugged into the APC powerbar is just as noisy.

Bottom line... The Belkin PureAV PF60 has a nice voltage/current display up front, is well built, it's convenient to have 13 outlets at one's disposal, and it has surge protection (unlike the Synergistic set I measured). I'm not looking to replace this unit any time soon. Remember that hi-fi gear needs good, well regulated power supplies that reject noise and it's not surprising that a good DAC like the TEAC doesn't benefit from any further power conditioning. I assume good DACs should perform similarly. Maybe there would be a more significant effect with cheap wallwarts, who knows...

Subjectively, likewise, I didn't hear any difference whether the TEAC was plugged into the PF60 or the inexpensive APC powerbar listening to Simon & Garfunkel's Bridge Over Troubled Water through the TEAC DAC.

As usual, I would love to see some measurements: real-world demonstration of power conditioning making a significant difference to a good DAC's noise floor for example.

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Been listening to more multichannel music lately. The audio high point of the week was being surprised the other night by how good the multichannel mix for The Carpenters' Singles 1969-1981 SACD (2004 release) sounded! Proper placement of Karen Carpenter's voice up front in the center, good balance of front-to-back volume. For an example of a poor surround mix - Neil Young's Harvest (2002 DVD-A); what's with the surround channels sounding unnaturally loud?

Time for some weekend R&R :-). Off to see Wynton Marsalis tomorrow night... Enjoy the music everyone.

MUSINGS: Silence (Is Golden)... [HTPC Rebuild]

A couple weeks ago in the post on the Philips Golden Ear Challenge, I touched on the topic of hearing acuity and the importance of this. It's one of those things that seems to be taken for granted in the press and in audio blogs/reviews as if everyone who writes on high-end audio is capable of these feats of perception.

This week, I thought it would be useful to consider the topic of "silence". Music grows out of silence. Without silence - or to more accurately put it; a low noise floor in the listening room - it would be difficult to detect very slight "microdynamic" changes. Even if one were to pump up the volume, nuances can be missed. In part, this is why the "dreaded" dynamic range compression (volume compression) is used. It reduces the dynamic range such that even very "soft" detail is pushed up in volume allowing detection of these details on the subway and in cars (remember back in the day when we had "loudness" buttons on car music players?), as well as qualitative psychoacoustic preference to some extent. There is a limit to how far volume can be pushed in that at some point, we experience the sound to be intolerably loud or the hardware starts distorting - remember to always protect your hearing. Important characteristics of accuracy in reproduction - tonal neutrality (uncolored), and precise conveyance of detail (combination of good dynamic range & timing accuracy) - demand that the room be isolated from external noise as much as possible. It'd be a shame to listen to high quality audio at reference 75-80dB level but 50dB of that is affected by noise! Even worse than consistent background hissing, humming, or rumbling is random or episodic noise like frequent cars passing by or people talking outside distracting the virtual "concert".

Refer to the "Sound Pressure" Wiki page with the relevant levels at the bottom. For convenience, I've reproduced it here:

Not on the list: AT&T-Bell "Quiet Room" = 10dB(A). Orfield Labs "quietest place on earth" as per Guinness = -9.4 dB(A)!

Nobody (that I know of!) advocates listening to music in an anechoic chamber of course. In a domestic sound room environment, your best bet for a quiet room would likely be in a basement behind closed doors unless one listens in the dead of night away from street car noise and domestic hustle and bustle. The professional standard for ambient noise is usually around 20dB(A) for the recording studio (check out this EBU Tech 3276 document for the gory details). As indicated by the red asterisk in the chart, we should try to aim for a very quiet 20-30dB SPL in the listening room; similar to the environment that the pro sound engineer would use as the reference.

In my home, as I mentioned last month regarding the HTPC in my sound room, I could still hear the hard drives spinning in that computer. A bit annoying, and this just won't do :-). So I decided to rectify the situation doing what I suggested in that post - separating the music/movie/data server component to another room and putting together a relatively low cost, less powerful computer which could act as a streamer. I extracted the fanless power supply from that computer and reinstalled the SSD with Windows Server; moving it into an adjacent room. Here are the pieces then for the new build:

Case: Bitfenix Prodigy M microATX
Power supply: transplanted the fanless SeaSonic SS-400FL2 400W
Motherboard: ASUS B85M-E/CSM - has HDMI with 4K capability
CPU: Intel Pentium G3220 (dual core, 3GHz, 54W TDP only, Haswell graphics features but slow 3D)
CPU Cooler: CoolerMaster Hyper 212 Plus (total overkill but lets me run almost fanless!)
SSD: Corsair FORCE 240GB (got a good deal on a refurb)
RAM: 8GB Kingston DDR3 1600 (note the G3220 will underclock this to 1333)

I installed Windows 8.1 Pro x64 on the system. Due to the oversized CPU cooler, I'm able to run the 120mm fan essentially silent between 20-30% speed and this is the only fan in the whole unit. No problem running 50 iterations of IntelBurnTest without any errors at "Very High" stress level while staying cool (I like using this program for stability testing more than Prime95; generates lots of heat within minutes).

So far, I've streamed a couple of MKV 1080P movies in excellent quality 20Mbps H.264 plus DTS soundtrack - no problem at all. I also installed JRiver 19 and foobar as I described in that last HTPC article and have no problem streaming DSD64/128 to the TEAC UD-501 using USB2 and 5.1 multichannel FLAC to the Onkyo receiver through HDMI. The Squeezebox system (Transporter, Touch, Boom, Radio) connects directly to the Windows Server machine and has nothing to do with this HTPC.

All this is running silently with a reasonably fast machine for media playback purposes. I should be able to play 4K as well using the built-in Intel graphics off the ASUS motherboard's HDMI 1.4. I tried streaming some YouTube 4K videos and they looked great on the 1080P screen - they're decoding reasonably well without much framerate issue at least so it'll be interesting to see how smoothly (or not) they play on a native 4K panel one day assuming I'm still running this rig...

Out of interest, I decided to try measuring the background noise in my sound room using a calibrated Behringer ECM8000. Realize that this inexpensive measurement microphone is not meant for low noise purposes with a self-noise in the low 20dB range (according to this link) mainly related to the small microphone diaphragm. I'm certainly not about to spend something like $2000 to buy the AcoPacific PS9200KIT for this purpose (this can measure down to ~8dB(A)). I figure if I can get a rough estimate, it'd be good enough. So, using a Radio Shack digital SPL meter (which only goes down to 50dB) for quck'n'dirty calibration for the Behringer with REW, then letting the Behringer measure the "silence" at the optimal listening position, I'm seeing this:

Quiet room, 10:30PM: HTPC/pre-amp/monoblocks/subwoofer/TEAC DAC/Transporter/room EQ DSP all turned on.

Not bad. It can dip down to the 28's and up to ~31 over the course of a few minutes of measurement. Good enough as a ballpark estimate aiming for the 20-30dB target. Using C-weighting (which is a more linear response profile vs. A-weighting which corresponds to human hearing), I'm seeing ~33dB(C). Prior to this new HTPC build, I was seeing about 35dB(A) with all those hard drives spinning.

Even though the ambient noise level is low, my room still has not been treated with acoustic panels so the room reverberation time remains a bit high. EQ'ing has provided a reasonably flat response described previously (+/-5dB around the target Brüel and Kjær "house curve" with recent digital EQ tweaks). Absorptive acoustic panels remain on my radar screen - I'm still contemplating aesthetics.

Happy listening everyone. Make sure to take a minute and consider the "sound of silence" in your audio room...

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I want to end this post on a more serious note (as much as I enjoy the topic of audio, it's only a hobby after all!)... I want to send my regards to Matt Ashland, the CTO of JRiver, the principle developer of Monkey's Audio (APE format - probably the most space-efficient free lossless compression system), and contributor to the DoP protocol (DSD over PCM). As some of you know, he had a fall in January and required surgical evacuation of intracranial bleeding; still recovering in hospital. I had the pleasure of exchanging E-mails with Matt last year around the time of the beta JRiver 19 release regarding the PCM to DSD transcoding algorithm, DST decoding, some bug fixes to JRiver, and his summer vacation with his kids. A truly genuine, generous gentleman and one of the unsung heroes of the computer audio hobby... My thoughts and prayers are with you and the family, Matt. Get well soon.

Matt has a CaringBridge page.

ADDENDUM: (March 7, 2014)
For those wondering, even with the Pentium G3220 underclocked to 2.5GHz, undervolted by -0.1V to consume <30W under full load with IntelBurnTest (~36C after 30 minutes), there is no problem upsampling 24/192 PCM to DSD128 in realtime with JRiver 9. CPU utilization <30%. I had to increase the TEAC DAC's ASIO buffer to 250ms to prevent some buffer issues though. Sounds great...

Given the price and performance of this CPU, I can't imagine any reason to build anything less powerful these days. I think it would be no problem running this fanless with a good sized heatsink if I underclocked it even further; 2GHz dual core with lower voltage fanless probably would be totally fine!

MUSINGS: Saturday AM B&W Nautilus Listening

Under immense market forces and thin margins for the majority of consumer electronics, it is no wonder that the specialist brick-and-mortar stores have gradually disappeared over the years. Big box stores are everywhere these days it seems (Best Buy and Future Shop here in Canada), and the proliferation of internet "stores" like Amazon has clearly taken centre stage for the consumer on the prowl for the best deal available. In some areas like the AV receiver market, I believe market forces have clearly resulted in marked improvements over the last 10 years in sound quality and meaningful features representing great value (think HDMI support, room correction, proliferation of lossless formats like TrueHD and DTS-HD MA). For audiophiles, I believe the main hardware developments in the last 10 years have to do with the quality of external DACs and software for better integration of computer audio into the sound room. The ascent of Class-D amplifier technology has been substantial as well but I doubt many audiophiles see a need to migrate to these amplifiers purely by virtue of this characteristic (unless of course you need higher efficiency, smaller size, cooler operation, etc...). Likewise, speaker technology has advanced especially in the material sciences used for drivers, but I think it's hard to justify new purchases based on this factor since there are so many other variables in the overall sound quality (like your room!).

I don't blame the small audiophile stores for pursuing to sell gear that have big mark-ups. Things like cables easily come to mind and it's certainly in their interest to promote these products and give them ample "rack space" for potential customers to peruse especially if there are manufacturer incentives - I just hope there is balance between the concept of aesthetics and sonic performance being presented. Likewise, magazines have to sell copy and attract advertisers, what other option do they have? And you can bet again the power of the manufacturers/advertisers in promoting products even if said product is of unlikely benefit for the customer. Without adequate financial viability, there is no business.

In Vancouver, there are probably only about a handful of hi-fi stores left where you can even have a hope of sitting down to listen in anything remotely resembling a decent listening room. Thankfully they still do exist and on occasion, it's nice to visit to check out (and of course purchase) the products, perhaps demo the exotic gear, and interact with knowledgeable salespersons (ok, salesmen - who am I kidding?!) who clearly show a passion in the hobby and in what they sell.

So, on a relatively cooler Saturday morning last week, I headed over to Hi-Fi Center (along Seymour just north of Dunsmuir) to have a look and listen to one of the most recognized speakers ever. I, like many of you probably have seen pictures of the Bowers & Wilkins Nautilus speakers over the years but had never actually heard one. 2013 was the 20th year anniversary of this flagship product from B&W so they've been doing a tour through a number of cities since last year and it was finally time to hit Vancouver. The well done presentation with Q&A was made by Murray Cardiff, Canadian National Accounts Manager for B&W.

Now that's an interesting looking speaker - in "Maserati Gray". WAF was alas poor for me but I suppose YMMV.

For those who may not have looked into the specs and history, here's the deal with the Nautilus (I haven't seen any formal detailed review/measurements to date):

- First released in 1993 - no substantial change to the design / sonic characteristic since. Check out this video, and part two for more (also check out the epic background music)! I don't know how many speakers can be said to not at least have a "Mark II" by now. According to Murray, about 4 units are produced per month for sales worldwide; not surprisingly many to Asian countries. Manufacturing facilities in UK and China.

- Current list price in Canada - CAD$70k a pair (includes external crossovers discussed below). I wonder whether there has been any price appreciation; I think the list price at release was ~USD$60k.

- The shell of the speaker is made of a composite material. Each complete speaker package is about 200lbs based on the specification sheet - evenly split ~100lb for the speaker itself and the granite base it's "standing"/bolted on. They stand (only) about 4 feet tall.

- 4-way driver configuration - 12" (300mm) bass driver (with characteristic mollusk shell spiral tube), 100mm lower mid, 50mm upper mid, and 25mm tweeter. Notice the tapered "transmission line" behind the upper 3 drivers. They're absorbent-material filled terminating with a small opening in the rear. Frequency response widely quoted as -6dB from 10Hz to 25kHz. Crossover points: 220Hz, 880Hz, and 3.5kHz. The speaker impedance is rated 8-ohms, no specification provided for sensitivity (somewhat meaningless anyhow in a set-up like this with external crossovers and separate amps as discussed below).

- All driver domes made of aluminium. This gives away the >20-year old design from the perspective of material sciences.

- There are no internal crossovers for the 4 drivers. You see the non-detachable blue speaker cable coming off the base. Another clue to the 20-year old design is just how thin the cable appears to be for something that contains 4 pairs of wires; a pair for each driver. The wires themselves are multi-stranded silver, and examining them closely, I don't think they're thicker than 16AWG and come in a standard 10' length (30' length also available). Of course this is technically not a problem at all for such a short length of speaker cabling but might surprise those who expect garden hose sized wire gauge for high-end gear!

Bare silver multi-stranded speaker wires connecting amplifier to speaker (red & black).

- External active crossovers connected between the pre-amp and amplifiers. I presume the technology inside hasn't been upgraded since 1993, so not likely DSP-based digital crossovers. (In fact I hope not given the state of digital converters back in 1993!)

For the demonstrations, Murray used a combination of standard 16/44 and some hi-res played off a MacBook Pro through a full Classé (owned by B&W Group) pre-amp / stereo amp rack:

You see the black 2-box active crossover just below the computer. Below that with the LCD screen is the Classé CP-800 preamp which functioned as a USB DAC for the MacBook (it got a good Stereophile review, ~18-bit measured dynamic range, very low jitter). Below are 3 stereo class-D CA-D200 amplifiers, each 200Wpc powering the tweeter, upper and lower mid drivers. Obviously that stacking arrangement highlights a major benefit of the class-D amps - they stay cool. Finally below is the large class-A/B CA-2300 amplifier delivering 300Wpc into 8-ohms for the Nautilus woofer - I noticed a good sized fan on the back which stayed quiet.

Unfortunately, I was not able to play a CD I had brought with familiar tunes. Nonetheless, there was a good selection of music on the Mac already and the demonstration ran through a selection of familiar music - Paul Simon (from Graceland), Sting (from The Last Ship), Aaron Neville (from Warm Your Heart), Lang Lang (Rachmaninov No. 2), Rose Cousins (Canadian girl from Halifax), Beatles, Daft Punk ("Lose Yourself In Dance" off Random Access Memories), Oscar Peterson (We Get Requests). There were also a number of classical pieces I wasn't familiar with.

As usual with these demos, it's very hard to judge the sound given the importance of the room and the music used. On the whole, it sounded very nice. Good frequency reproduction from top to bottom. The upper end was very detailed and it was easy to discern low-level details including tape noise on analogue-sourced tracks and things like musicians taking breaths during classical performances (if you listen for such things!). I'm still a believer in a good subwoofer to tighten those lower registers however. Unfortunately, no hip-hop or rap was on offer in the demo so it was difficult to determine just what isolated beats down at 20-50Hz would have sounded & felt like. Another nice quality was the fact that the system could be pushed to high levels in a moderate sized room without a hint of strain with peak transients.

I was struck by the soundstage. The stereo effect was good even somewhat off center and this set-up certainly conveyed good horizontal and depth dimensions. I've seen comments from Nautilus owners describing excellent sound due to minimal rear reflections and folks placing these speakers against walls and other reflecting surfaces. Some of this may also be a reflection (pun intended) of the external active crossovers doing an excellent job with subtle parameters like time/phase alignment.

I heard a few of the attendees mentioning that the sound was fatiguing... Well, what can one expect when a good amount of the demo music was dynamically compressed?! Seriously, IMO, if these speakers did not convey the harshness of some of the tracks, then there'd be something wrong since that's just what the recordings sound like. Having said this, I did wonder if there was a bit of "ringing" from the aluminium tweeter which may accentuate the fatigue effect in some of the tracks (especially synthetic material like the Daft Punk)... I cannot be sure of this, just an impression. Obviously B&W has advanced over the years to using diamond tweeters while other companies like Focal, TAD, Magico, Ravel and Paradigm Reference are using pure beryllium to achieve higher breakup frequencies significantly above the audible spectrum. I suppose the use of class-D amplifiers could have an effect (likely not) - just don't tell me silver speaker wires sounds "bright" unless there's evidence of such a thing :-). B&W recommends at least 100W for these drivers and given the amplifier demands, I cannot imagine many folks running these with tube amps. I suppose NOS DACs and various upsampling DACs with rolled-off filter settings could tame the upper end at the expense of accuracy of course.  As I mentioned earlier, I have never seen a full review on these speakers and would be very curious what measurements look like.

The System

After the Nautilus demo, I got to wander around the store to check out some Totem speakers, and had a listen to the D'Agostino Momentum monoblock amps (yes, very pretty!). Also managed to put my test CD through the paces with this B&W 802 Diamond system:

As you can see, it's powered by a McIntosh MC302 300Wpc stereo amplifier. SACD player was also McIntosh. Really nice sounding as well, definitely goes deep although I noticed some low bass accentuation in that room (sounds nice and punchy but I would have used some EQ to tone down the bottom end a little). I had a listen to these speakers last summer in Singapore at The Adelphi, coming away with a similar impression of the excellent sound quality. I'd certainly be very happy with these at home :-).

Okay, time to wrap up! I'd like to thank Hi-Fi Center again for the opportunity to listen to the Nautilus - a classic icon of the audiophile world. To be in continuous production for >20 years in these days of perpetual upgrades and engineered obsolescence is quite a feat for anything; much less an electronic device. This opportunity is also a testament to the value of having good "brick and mortar" specialty stores where the knowledgeable, enthusiastic and friendly staff can answer questions and provide informed suggestions. Sadly, an almost extinct species in this day...

MUSINGS: Golden Earism (& The Philips Golden Ears Challenge)

Don't you love the term "golden ears"?

I wonder historically when this term was first coined. I suppose it must have been in the murky distant past of times immemorial when primitive man glazed upon the yellowish gleam of Keynes' "barbarous relic" and began ascribing all manners of idealistic properties. Or not...

The other day, a forum poster brought up this link from What HiFi? about Hi-Res audio. Yeah, it covers the basics, but I did want to add an item #5 in terms of "factors to consider":

5. You must have good enough ears to appreciate the difference hi-resolution makes.

I know this can be touchy for some folks, but it is what it is. Our ears (and brains), like all the other sense organs (and cognitive domains) do not have infinite resolution. And like everything else, time is not on "our" side. At 42 years old this year, my ear's "frequency response" only goes up to about 15-16kHz at normal amplitudes. Do I really have the need to go for files with sampling rates of 88kHz+? Honestly, I don't think so... But as I've expressed elsewhere, this is about perfectionist audio so I'm certainly happy to have access to my favourite music using the most accurate technology available (I'm still of the opinion that 24/96 is more than I'll ever need in terms of the technical specs).

If you haven't seen it yet, recently, our friends at Philips have come up with a very cool website called the Golden Ears Challenge. Just enroll with your E-mail address and get going with some ear training. Log off and it'll keep your place in the test. Seriously, if you believe your equipment and ears are up to the task, take the challenge! I suspect some audiophiles will be surprised at the limits of their hearing ability.

I took the Challenge using the ASUS Essence One on my desktop with a pair of venerable Sony MDR-V6 (<$100) studio monitor headphones. I figured, if the V6 is good enough for Roger Waters, it's good enough for me!

I suppose better headphones like my Sennheiser HD800 and being in my much quieter audio room downstairs could have made tasks like hearing high frequency extension or detection of minor amounts of reverberation easier, but the computer desktop was more convenient... Remember to make sure the DAC is set to native 44.1kHz and something like Windows Mixer isn't upsampling.

The Bronze level wasn't difficult at all unless one has hearing issues, I suspect.

Silver level was achieved in one sitting. I think the test music samples were encoded with 320kbps MP3 and it wasn't too hard to differentiate between 320kbps and 128kbps lossy compression - as usual, listen especially to the treble and see if you can hear the loss of detail, more "brittle" rendering, and slight "chirpiness". I found it more challenging detecting small amounts of reverb down at dry/wet ratio of 0.15 later in this test.

Here's the "coveted" Golden Ears achievement :-). You get an E-mail to confirm.

Not bad, took my time over a couple of nights in between some virtual paper work. Achieved with a little patience and using the same ASUS Essence One / Sony MDR-V6 combination. The most time consuming part was getting the Boost/Cut Identification Test right at the various frequencies (63, 125, 250, 500, 1k, 2k, 4k, 8k, 16kHz). A frequency boost at 16kHz with real music was barely audible for me. Tests like the bass boost really benefits from headphones capable of good bass response so I suspect an open unit like my AKG Q701 would not be the best headphone to try this on.

As of this writing, here are the overall statistics for this test:

Not bad, looks like I'm one of 117 who have completed the Golden Ears level so far out of 925 who finished the Basic level (12.6%). Objectively, Golden Ears aren't that rare :-).

Anyhow, I highly recommend giving this a try yourself... I believe that anyone can have an opinion about equipment fidelity just like everyone has the right to have an opinion on what music they enjoy. But it does require good ears technically if one is to claim discernment of small differences between pieces of gear. I think for many, engaging in tests like this one would be very educational if not eye opening in terms of limits of one's hearing. Furthermore, I think doing challenges like these should be mandatory for those who engage in "professional" audiophile hardware reviews focused on audio fidelity. (And those championing technical specifications based on 'articles of faith' like 24/192 over 16/44... errr... who's that Pono guy again?)


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PS: Remember that JPlay software I measured awhile back which made no difference (actually there was a bug in that version which makes it even worse)? Looks like they've returned with a line of "JCAT" hardware! USB cable for 299Eur, SATA for 349Eur! How about Cat 5e (!) for 349Eur - man, they didn't even bother trying for Cat 7; at least the AudioQuest "audiophile" ethernet cables did! Reminds me of sellers on eBay trying to scalp some bucks by listing items at huge mark-ups with "Buy It Now" to catch shoppers who have never tried "The Price Is Right". This time around, they don't seem to claim sonic superiority of these products - merely "help you create the ultimate PC audio transport and get the most out of JPLAY".

Considering that a high quality SATA-III cable with fastening clips (which this one doesn't even seem to have!) runs for about the equivalent of <$4Eur at my local computer store, these guys are charging at >8700% mark-up presumably for that JCAT logo stamped on, silver plating and teflon coat (of course unless you're pimped out and have a window into your computer case, these will be hidden from view)... Sorry J-Dudes, but IMHO, The Price is Gluttonously Wrong. Why don't you guys show us in what way these are better than good quality generic SATA-III 6Gb/s cables first? (As if there's even a plausible explanation.)