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.)

As you know from previous posts, I'm not a DSD fanboy. It does have some limitations compared to PCM; the technological limitations themselves in terms of high frequency noise due to noise shaping, non-uniform noise floor as a result, lack of opportunity to run DSP algorithms, and the current file format implementations make it cumbersome. In business, the opportunity to differentiate oneself from another provides the opportunity to sell the item as "new", "different" and "better" and indeed DSD provides many talking points and sales opportunities. As I wrote in the PCM-to-DSD article, there are certainly some audible differences DSD processing imparts and this change can be perceived as euphonic even though the actual underlying resolution is no better. It's a bit of a philosophical point: is it better to listen to something played back accurately? Or should we aim for euphonia even though it clearly adds something (distortion) to the signal that was not there originally? In the case of the PCM-to-DSD algorithm, clearly ultrasonic frequencies are being added to the signal as demonstrated by the measurements. As I have opined a number of times in the past, my preference is for accuracy; this is my definition of high fidelity and I generally feel that PCM retains accuracy better than DSD64 and with DSD128 obviously capable of better accuracy than DSD64. (Of course, if a recording began life as DSD and wasn't tampered with, that's as good as it gets and no point losing precision going to PCM...)

Let's consider in this post what it would take to overcome the "limits" of 16/44 in a home component system. Remember that headphones are a much simpler case and for a lot less money, one should be able to better speaker systems - but of course the presentation of soundstage would be much different. (In a way, this post will be similar to a previous one I linked to in What Hi-Fi? but I hope to be more thorough and more realistically critical of the whole endeavour.)

I. The High-Resolution Hardware Needed.

A. Good enough DAC?
Let's start with the DAC since unless it can reproduce analogue waveforms of high dynamic range and the full frequency spectrum, there's no point proceeding. I have already posted measurements throughout this site and demonstrated these qualities in many DACs already. I do not believe it is difficult to pick out DACs which measure well. A DAC capable of >16-bit dynamic range is not hard to find; almost any decent unit today should do (even the electronics-packed Squeezebox Touch from years back can do this). Furthermore, it doesn't have to be expensive; that old AUNE X1 DAC (<$200) from at least a couple years back can easily convey the measurable benefits of hi-resolution PCM up to 24/192. If you have a look at the Stereophile measurements page for modern DACs, it's easy to see that the state-of-the-art DACs all essentially measure in an ideal fashion these days for 16-bits with the best achieving a noise floor around 21-bits with high-resolution material. Talk about 32/48/64-bits and such is nonsense in terms of DAC output resolution. High bit-depth would be beneficial for accuracy of complex DSP calculations like in the studio environment with numerous Pro Tools effects. Although I haven't heard all the "best" DACs, I have heard a number of good ones and feel that differences are minor and most likely inaudible in volume-controlled blind testing despite reviewers' comments. Measurements can show slight differences with impulse response, the occasional jitter difference. No biggie.

B. Good enough pre-amp, amp, speakers?
If you look at the measurements in Stereophile under the pre-amp section, it's not unexpected that vacuum tube preamps generally measure poorer in terms of unweighted audioband SNR; typically around 75-85dB with a 1V signal including some very expensive models. In comparison, good solid state preamps should be able to achieve around 100+dB with a similar test. In general, you'll find this difference with any vacuum tube vs. solid state comparison... As per the philosophical question posed above, vacuum tubes are less accurate (older obsolete technology) but can be euphonic (and nostalgic) for some people.

In my system, I've already shown that the Emotiva XSP-1 pre-amp and Onkyo TX-NR1009 receiver are capable of passing through an analogue signal of easily >100dB dynamic range from the TEAC UD-501 DAC. Unfortunately the old Denon AVR-3802 was incapable of this, therefore, it would not be a device I'd use in the high resolution audio chain. Measurements are needed to know whether the gear is good enough for high-resolution audio.

The amplifier and speakers are even harder to quantify... For high resolution dynamics, we need the amplifier to be able to produce enough power to drive the speaker to create >96dB dynamics in the room! Likewise the speaker will need to be able to handle the power to recreate the full dynamic range without distorting to a significant degree in order to benefit from the resolution afforded by >16-bits! This is tough.

That's just the dynamic range. What about the frequency response from a speaker? Assuming that ultrasonics have some kind of beneficial effect (dubious), most excellent speakers can reproduce frequencies beyond 20kHz. Few speakers have a "super tweeter" to reproduce those frequencies up to 40+kHz flat however. In fact, many listeners would prefer a slightly rolled off response by 20kHz as per the Brüel and Kjær "house curve". I know I can't hear much above 16kHz so would find it hard to feel any need to spend money on accurate ultrasonic frequency reproduction.

Success in conveying the high resolution audio signal also depends on...

C. Good enough room?
It goes without saying that to achieve a dynamic range over 96dB, we need a quiet room; the main point of my previous post about the importance of "silence". Assuming you have amps and speakers capable of it, to better the dynamic range afforded by the 16-bit CD format in a very quiet 20dB(A) listening room, we need to be able to achieve dynamic transients >116dB SPL. In my stereo system consisting of 250W (continuous power) monoblocks feeding quite sensitive 92dB/W/m speakers, placed near a wall, the theoretical maximum SPL at my sweet spot 11-feet away is "only" 111.5dB with the amp already contributing 0.05% distortion (check out the Collins' Cinema SPL calculator to do your own calculations). Accounting for some headroom, I can maybe get up to 115dB for dynamic transients in music. Remember, due to logarithmic properties, even if I doubled the amplifier power to 500W, the SPL would only increase by 3dB to 114.5dB (maybe getting close to 118dB transient peaks). I suspect the speakers would be significantly distorting the sound at these loud volumes, not to mention we're well into hearing damage levels with repeated exposure.

Furthermore, in-room frequency response, reverberation time, speaker placement effects are also significant... I believe all of this would be of greater effect than what is afforded by the extra resolution.

As you can see, we've got a problem here already from the hardware setup perspective in arguing for high resolution audio of >16-bit dynamic range and >22kHz frequency response.

[Note that I haven't even mentioned dithering and noise-shaping increases the perceived 16-bit signal's dynamic range even further at reasonable volumes, so in fact, it's not as "easy" as the 96dB quoted above. Read this iZotope Ozone guide for more details.]

D. Good enough ears/brain?
Serious golden ears (platinum ears?) are mandatory I believe :-).

 

II. The High-Resolution Software Needed.

When we buy our music, how do we know that all the complex bits and pieces were done well enough to ensure quality? Furthermore, when we go on a web site like HDTracks, Qobuz, 2L, Channel Classics, etc... how do we know it's worth downloading 24/48, 24/88, 24/96, 24/175, 24/192, DXD, DSD64, DSD128?

We don't.

Sadly, I think the state of affairs right now for many audiophiles (promoted by music web sites, distributors and those with financial/advertising revenue interests) is akin to the megapixel race in digital cameras. Thankfully, that megapixel race seems to have died down significantly as consumers have gotten wiser to the fact that quality of each pixel is important. The JPEG image from a cellphone's 12 megapixels is way inferior to that from an SLR with nice lens at 12 megapixels (even though JPEG itself is lossy)! But in audio, there's still the impression that bigger numbers are somehow supposed to be better without qualifying where the source comes from. 192kHz is better than 96kHz. DSD, the "super audio" format beginning at 2.8MHz sampling rate is believed to be even better (by some).  I have yet to see a professional reviewer prefer the sound of a 24/96 over 24/192 when both versions are available - is it because he has hearing higher than 48kHz? Do all DACs sound better running at 192kHz? Or is it just bias from the numbers game? (I suggest it's just the latter.)

Of course, in real life it's never as simple as bigger numbers being better... As consumers we usually have no insight into the hardware used like the microphones or how the ADC was accomplished (remember the technical limits of the studio equipment!). Complex studio "processing" for multi-tracked projects, and the myriad of DSP plug-ins require expert sound engineers to ensure quality with each step. I would be remiss to bring up concerns around decisions to use dynamic range compression (eg. Loudness War) which could have artistic merit but which also lowers the ultimate fidelity of the original recorded signal and diminishes the number of bits of dynamic range required to fully encode the sound. As a result, many audiophiles including myself would regard some of the first CD releases back in the 1980's to about mid-1990's as superior to subsequent remasters using heavy handed processing despite the fact that ADCs back in those days would have been inferior. For example just look at the mess UMG did in 2010 with the Rolling Stones remasters among many recent cases.

To some extent we could make decisions based on the reputation of certain companies. Audiophile remastered releases by Analogue Productions, Audio Fidelity, Mobile Fidelity can generally be accepted as the best remasterings using better equipment and maintaining higher standards of audio quality whether it's standard CD, SACD, or potentially high-resolution PCM. Some mastering engineers like Barry Diament, Kevin Gray, and Steve Hoffman are among those who have made a name for themselves as mastering to a higher audiophile standard (as opposed to this guy and his ruinous work on RHCP's Californication and I'm With You). Realize of course that remastering demands that the original source material to be of high resolution quality if one is to show the benefits of the high-resolution format over CD - this is debatable for even the best analogue tape. Likewise, certain record labels like 2L, AIX/iTrax, Channel Classics, Naim, Blue Coast, Reference Recordings produce superb sounding original recordings catered to the audiophile segment with a strong reputation to maintain.

However, we are witnessing the rise of the high-resolution internet store fronts. Places like HDTracks, Qobuz, Acoustic Sounds, Native DSD Music, Linn seem to be resellers of high-resolution format files (PCM and/or DSD) from some of the major/larger record labels. Quality control issues have been discussed over the years as obvious "upsampling" (taking a standard 44kHz and resampling it to 96kHz to sell for example) have shown up on places like HDTracks which really does a disservice to the music buying public. Like the list of SACD's appearing to be nothing more than upsampled PCM, we do not know the origin of many of these "high resolution" albums. I've certainly run into my share of questionable releases that appear to be blatant standard resolution upsampled music or looking like standard 44/48kHz PCM data run through an analogue board and re-recorded with low-level high-frequency noise on a 24/192 release. The former is easy to spot, but the latter is hard to prove!

As audiophiles and audio enthusiasts, we can of course look beyond just the hyped numbers and acronyms like 24-bits, 96kHz, DSD... I suggest doing a search on places like the new Computer Audiophile Music Review segment (good job Chris on the new feature) and see if folks have "measured" the release for the following characteristics to look for a "true" high resolution master as best we can as consumers:

A. Absence of steep low pass 'filtering' suggesting upsampling.
Good examples are all over my list of suspected upsampled SACDs. Note that in the case of DSD, since there's rising high frequency noise, the discontinuity exists as a steep cliff around 22kHz. You can also see a good example of this in the HDTracks release of Graceland 24/96 as per this thread on Computer Audiophile. Because upsampling of PCM is done within the digital domain, it results in the "brick wall" abrupt transition like this:

Blatant example of 24/96 upsampled to 24/192 using "Blue Train", John Coltrane. No frequencies above 48kHz at all...

Since this is too obvious, and analogue tape always has high frequency noise, let's just add a little bit of low-level -100dB white noise above 48kHz which would be inaudible:

Upsampled track with -100dB spatialized white noise mixed in to look like there's something >48kHz.

There, you'd never know this was just an upsample by looking at the spectrum (but can perhaps guess if looking at the spectrum in realtime playback).

Here's what that track in true 24/192 looks like, but as shown above, it's really easy to conceal a 24/96 upsample:

The real 24/192 "Blue Train" off the 2001 Classic Records HDAD release.

On a side note, one experiment to try is to take a high pass filter of everything above 30kHz on a 24/192 file and bring it down in tone into the audible range and have a listen... I rarely hear anything correlating with the underlying music - this is a big reason why I rarely bother with 24/192.

B. High average dynamic range.
What is the point of going above 16-bits if the mastering is compressed to hell and back? The opportunity to easily measure the DR "Dynamic Range" value using the foobar plugin (use the newer version found in the DR Database) has been extremely useful. This value is calculated by looking at the difference between peak volume vs. RMS "average" volume over chunks of time throughout the music. A higher number correlates to the presence of loudness peaks which of course corresponds to a more dynamic sound. Real life sounds dynamic! You can also check the Dynamic Range Database before purchasing to see if the title is in there.

Although there is no simple rule about this, the DR value can give you insight into whether the high resolution version of an album is the same mastering as the CD release and what difference exists. Consider the example of Lorde's Pure Heroine...

CD Release:

Here's HDTracks 24/48 "Studio Master":

In this example, the HDTracks release looks (thankfully) a little bit better than the CD - this is actually somewhat rare with modern pop/rock recordings since I find the majority are the exact same master with the same DR value. Generally we're seeing a 1dB improvement in average dynamic range with the high-res version. The problem is, we're still at a DR of 7dB only! Furthermore, there are no nuances to the peaks with every track pushed up and limited to 0dB (not really an issue here since the music is clearly synthetic but this would be very bad to see with acoustic music or orchestral music like some soundtracks such as the recent DR7 The Dark Knight Rises).

For perspective, back in the late 1980's and early 1990's, the average pop and rock album had average dynamic range around 10-13dB. A well recorded classical or jazz album with all the dynamic transients that come with natural recordings should average around 13-16dB.

Personally, my audio "New Year's resolution" for 2014 is to avoided purchasing *any* high-resolution album without at least a value of DR10; and I feel this is very conservative already (I probably should advance this to DR12). The rationale is simple...  Recordings with low dynamic range and peaks pushed to ~0dB are loud and do not require us to turn up the volume to listen at reasonable levels - typically I play DR7 tracks at something like -35dB through my system. This type of recording does not utilize anywhere near the limits of a 16-bit medium like the CD nor that provided by my home audio system, much less demand the 24-bit dynamic range of high-resolution downloads.

As an aside, be very cautious in using the DR Meter for vinyl rips! Just because the values tend to be higher doesn't mean that it's due to the LP having better mastering. Have at look at Ian Shepherd's video here (essential viewing IMO):

Why You Can't Measure Vinyl With TT Meter

In that example, with pops and clicks removed, even though the DR score is higher with vinyl, the original source is the same and we can speculate in a later post as to why this might be the case. Remember that vinyl rips remain plagued with the limitations of the LP source including surface noise, subtle wow/flutter, inner groove distortion, idiosyncrasies of the playback gear, etc. They can and do still sound great especially with some judicious post-processing, but not true high resolution like a good quality pure digital recording.

Nonetheless we see on DR Database someone's vinyl rip of Pure Heroine:

DR10... Not sure if that is an actual better master or just the effect of transfer to vinyl (as per Ian Shepherd's example). With only a 3dB increase compared to the HDTracks version, I do not believe we're looking at a true remaster effort here.

III. Conclusion - Expectations for High-Resolution Audio?

So, let's try to answer the question posed above of what to expect from high resolution downloads in conclusion... I think the answer is simple: Not much if anything compared to a technically good 16/44 version or CD of the same mastering.

From a hardware perspective, digital audio technology has advanced to the point of easily overcoming the 16/44 "limitations" within the DACs. The problem revolves around the ability of the analogue side (especially the speakers and room) and ultimately, the perceptual limitation of being human. Apart from listening in unreasonable ways (eg. listening to fade outs at extreme volume levels comparing undithered 16-bits vs. 24-bits), I don't know of any controlled study suggesting an audible difference between standard and high-resolution music.

From the software perspective, we are faced with  just how well the recording was done and mastered in the first place. I would gladly listen to an MP3 192kbps of a well recorded/mastered album than a poor DSD128 or 24/192. For folks to place an emphasis on the value of 24-bits, 88+kHz, and DSD appears to be ultimately a fallacy. The most egregious examples of these are the pseudo-high-res albums like the upsampled SACDs or upsampled PCMs sold under the high resolution moniker. But as Mark Waldrep ("Dr. AIX") has been talking about for years, true high-resolution demands that the source be high resolution... The hardware reproduction challenges are difficult enough, but if the source is anything other than a pristine (digital) production, the potential resolution of 24/96 and above would never be utilized (assuming anyone can even hear it!).

I often criticize Neil Young and his comments on these pages (and I see many others do as well) simply because he hypes up the 24/192 number in a way personifying the "megapixel race" of the audio world (I'm sure he's a decent fellow and all, and I certainly respect his artistry... But just watch this cringe-worthy "Dive Into Media" video from 2012 to see what I mean.). I think it's quite clear that unless his Pono brings something truly new like getting the record labels to provide good quality, dynamic, remasters, there is essentially no hope of Pono hi-res sounding any different than what we have so far with HDTracks or Qobuz. The 16-bit, 44kHz resolution "container" was never a significant "problem".

In the years since my post in 2010 (Appendix A), my opinion really (perhaps to my surprise) has not changed. Even with better equipment like newer DACs, better headphones (Sennheiser HD800), a much improved room, separate components, better speakers. Taken together - the challenges of hardware and software - I suspect that high-resolution PCM (like DSD) doesn't have much future in "making money" for the companies other than as it is today... Niche products with a small target audience. Unless the software companies somehow decide that the default digital resolution is greater than 16/44 and provides it as such, I don't see why the average person would pay more for what essentially is of little (if any) benefit in the consumer environment. The thought of making high-resolution lossless downloads default is unlikely given that the majority still just buys music from iTunes which isn't even lossless to begin with!

Now on a personal note, I am mindful that hobbies in general are not meant to be rational pursuits; they're emotional endeavors for the sake of pleasure and fulfilment of some form. We (audiophiles) probably spend more time than needed obsessing over all manners of "trivial" things from what to buy to whether a cable is "good enough". Most people in this world would look upon the perseverative audiophile as a form of obsessive-compulsive neurotic but so be it. So are car collectors, stamp collectors, wine connoisseurs, watch aficionados, etc. In this context as an emotional pursuit, I'm actually OK with still collecting my favourite music in 24/96 or DSD if that's "as good as it gets". It is about perfectionism. There still could be something wrong with brickwalling at 22kHz and 16-bits do not challenge the objective capabilities of my DAC - perhaps some day I'll hear the difference assuming there are some merits to the recording being "high-resolution". Nonetheless, to buy a recording which is just upsampled or has severely squashed dynamics offering no potential benefits at all to the collector is just plain foolish. As a "more objective" audiophile, there are limits to perfectionism which I would be unwilling to cross based on the scientific data.

For those who insist that they can hear high-resolution audio, I would love to see comments below... I would highly encourage doing a controlled test like the ABX below for yourself with your own high-resolution music. Let me know if you ABX successfully!

Recently, I've been listening to some Larry Carlton. Excellent stuff especially some of the older recordings from the early/mid 1980's - have a listen to Discovery from 1987. No argument from me that 16/44 sounds fantastic when done well!

Audio low point of the week? A friend notified me that Beck's recent album Morning Phase scores DR6 from the HDTracks 24/96 "Studio Master" (I see the Computer Audiophile just put up an article on this). Same as CD. As per my New Year's resolution above, this is one album I'll just pick up at the local CD vendor for $10.

As usual... Enjoy the music. I've promised my wife I won't be buying another copy of "Kind Of Blue" - even in High-Resolution Audio 32/384 :-).

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APPENDIX A: Old post from Audio Asylum in 2010

Test with 24/96 vs. 16/44 LITTLE TO NO DIFFERENCE.

Posted by Archimago (A) on February 13, 2010 at 09:00:59

As a "data point" and discussion for those who have been singing the praises of hi-res audio downloads, I did a test recently.
Recently got the very versatile E-Mu 0404 USB (AKM AK4396 DAC) to play around with on my computer (quad core 2.8GHz, 8G RAM, low DCP latency, Win7, through USB of course). With some recent high definition downloads / DVD-A source:

Rebecca Pidgeon - The Raven: "Spanish Harlem" (24/88 Bob Katz 15th Anniversary Ed)

Carol Kidd - Dreamsville: "When I Dream (2008)" (24/96 Linn Studio Master)

Laurence Juber - Guitar Noir: "Guitar Noir" (24/96 AIX DVD-A rip)

Took these FLAC/WAV files, down sampled in Adobe Audition to 16/44 (no dither, no noise shaping) then resampled back up to 24/96. Verified that frequencies all truncated to 22kHz. Then listened to them with Foobar 2000 ABX comparator using the E-Mu ASIO output plugin. This allows me to A-B on-the-fly and do some "blind" ABX'ing.
Listened with headphones: Audio Technica ATH-M50, Etymotics ER-4B.

With this setup, I figure I've removed all variables except for sample rate change - same mastering, same DAC running at same sample rate.
Results: Essentially NO DIFFERENCE between the native 24/96(88) and 16/44. Blind ABX results NO SIGNIFICANT DIFFERENCE. When I do the rapid A-B switch in the middle of a song, I thought there MAY have been slightly more smoothness/openness in the high-def version but this could just be placebo and the improvement was MAYBE 5%.

At 38 years old, very few loud concert experiences, I don't think I have 'tin ears' (hey my wife thinks I have better ability to pick out music in noisy environments so I guess it's at least as good as some females :-). 

My conclusions:
1. Either my equipment sucks or these samples suck and there's alot more but I need to fork up more $$$$.
2. Or high-def cannot be well appreciated with headphones.
3. Or the upsampling back from 16/44 --> 24/96 somehow reconstitutes the sound.
4. Or, there's really not much difference.
5. At this point I'd probably spend a few more dollars to buy a high-def download (maybe at most $5-10 more if it's something I like) when given the option but not expect significantly more revelation in the sound. 

I've listened to good SACD as well and like them but there's no way to do tests like this. I didn't bother with 24/192 material since I figured most improvement should come from this first step up 44 --> 96. Anyone else done such tests for themselves?

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 DDR 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.)