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ONKYO TX-NR1009 In The House...

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Happy Canadian Thanksgiving long weekend everyone! Sadly we don't have the same "Black Friday" deals you lucky Americans enjoy :-).

I realized something the other day... I refuse to be deprived of Dolby TrueHD and DTS-HD Master Audio (MA) in the new home theater / sound room! These are of course the "new" (DTS-HD MA announced back in 2004, TrueHD in 2005) lossless surround formats available on Blu-rays. Furthermore, I want to finally be able to play multichannel FLAC files created from DVD-A and SACD rips, of which I probably have about 100 albums archived away on my music server waiting for the day I can get a decent multichannel DAC (the recent USB exaSound e28 looks interesting). There is only one reasonable and relatively inexpensive way at this time - embrace the HDMI interface.

Two options: either get a new surround processor like the Emotiva UMC-200 and use the external inputs of the Denon AVR-3802 for amplification purposes, or just upgrade the receiver to something (much) more modern since the AVR-3802 was bought at a time before adoption of HDMI (way back in 2002!).

After much humming and hawing, I decided to have a look at the used market locally. Fortuitously, a minty 1.5 year old ONKYO TX-NR1009 was available at an excellent price! The reviews (here and here) were good and the features and build looked fantastic so I decided to plunk down some cash for this baby:


A multitude of I/O ports and connectors. I doubt I'd ever touch composite, S-Video, and component analogue inputs again!
Now this specific model was released a couple years ago - it came out in mid-2011. It's "THX Select2 Plus" certified, capable of 9.2 surround, has 9 amplifier channels (potential for passive biamping the front speakers), all the usual digital sound decoding capabilities from Dolby and DTS along with 7.1 LPCM from HDMI up to 24/192, Audyssey MultEQ XT and the capabilities of HDMI 1.4 for video (3D, 4K upscaling). This model has since been superseded by the TX-NR929 (I would have thought the TX-NR1010 would have filled this role based on model number but the NR1010 actually has 7 amplifier channels, rather, the TX-NR3010 is the higher end model with 9 channels). In any case, it looks like the newer 2013 generation has Audyssey MultEQ XT32 (note the extra "32"; more and higher resolution room EQ control), built-in WiFi and Bluetooth, AirPlay, and 4K passthrough - cool but not essential for my purposes. Compared to the stereo audiophile world where we argue about the audibility of PCM vs. DSD vs. minute differences between digital filters, the home theater domain brings with it more features than most music lovers would likely care to know... I suspect that as the technology continues to mature in the future, we will see stabilization of feature set and the need for upgrades will diminish for most end users; if this has not already started.

Speaking of the future, 4K video is on the horizon though its mainstream commercial appeal is far from clear. I wonder if the consumer digital video world will play out like audio - 1080P becomes like the "CD standard" and 4K takes the role of SACD/DVD-A/hi-res downloads. The 4K image improvements can clearly be demonstrated (go have a look at a 4K TV near you), but other than videophiles and those with >60" screens, it's going to be hard to justify the improvement for most TV sizes, at most viewing distances. Also, there's currently precious little media out (the Sony XBR TV I saw comes with a Sony PC loaded with some sample material). A new Blu-Ray standard needs to be formalized (see recent news about 100GB 3-layer BD). It's also unclear whether current Blu-Ray players can read these >2 layer disks; even if they can be read, I suspect the players might need H.265/HEVC decoding which likely means brand new hardware. One area I can see 4K could be beneficial to smaller screens is in synergy with passive 3D giving a full 1080P image per eye without the drawbacks of active 3D glasses (I've been using a 55" LG passive 3D TV now for 2 years so can attest to the resolution limitations)... However, 3D movies have not taken the world by storm so I'm not betting on this to fuel sales :-).

I digress... Back to the ONKYO and sound... So far, I've plugged in my HTPC to try out the HDMI input - sounds great off the AMD A10-5800K "Trinity" computer for both music and movies; nicely detailed and dynamic. Multichannel SACDs sound great (DSD converted to multichannel FLAC played back with JRiver). Technically the amplifier portion does have more power than the Denon AVR-3802 with decent measured wattage even with 5 and 7 channels driven. With all DSP off and in "Pure Audio" mode, it's more "weighty" than my old Denon regarding bass impact; less "forward" sounding. It actually sounds closer to my recollection of the Simaudio Moon i3.3 integrated amp but that's of course from memory which we all know could be inaccurate.

Anyhow, I'll try to run a few measurements on this machine in the next while when I have some time. I'd certainly be very curious what the numbers/graphs look like compared to other DACs tested so far. According to the spec sheet, the DAC consist of an 8-channel TI PCM1690 and stereo TI PCM1789; both with rated SNR of 113dB - I'd certainly like to see if they can achieve anything close to this in a compact full-featured box with all the digital processing going on plus 9 power amplifier sections! This would also be the first time I'll have the opportunity to have an objective glimpse at performance off a recent HDMI implementation.

Onkyo tattoos!

BTW: Even though the ONKYO will be the heart of the surround system, I still have the Emotiva XSP-1 which will form the basis of the 2-channel signal chain. One which I will take advantage of once I get some monoblock amps; likely in 2014.

Music for tonight:
Feargal Sharkey - Feargal Sharkey (1985), man, haven't heard "A Good Heart" for more than a decade! Also revisited another 80's memory: The Jitters "Last of the Red Hot Fools" (1987) - Canadian, eh?

Addendum: After writing the above about 4K last evening, I ran across this link: 4K Blu-ray is dead tech walking. Yup, sounds about right! :-)

MEASUREMENTS: Separate vs. AV Receivers (Emotiva XSP-1 vs. Denon AVR-3802 vs. Onkyo TX-NR1009) as Analogue Preamp.

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Okay, so maybe I'm being a bit too dramatic using that epic battle between Godzilla and Rodan above :-).

With the recent acquisition of the Emotiva XSP-1, I wanted to see just how well a separate pre-amp with audiophile design in mind stacks up against something more ubiquitous like the AV receivers out there. Remember that a preamp at its core has very basic functionality - it allows switching of the source and volume control by adjusting voltage gain on the output. The essential difference between a good preamp versus poor one (beyond features, ergonomics, remote control quality, etc...) is how well it maintains a high signal-to-noise ratio (SNR). If you supply a high resolution line level DAC output, you want to see that signal come out of the preamp with as much resolution as possible; this demands that the preamp introduce as little noise as possible.

Is there good evidence that spending money on a good analogue preamp will result in more accurate music reproduction? Let's find out in this installment...

First, let me introduce the contenders today:

Emotiva XSP-1:

Currently the highest quality Emotiva preamp out. The claim to fame is the differential design for balanced operation. It provides 2 balanced inputs along with a host of unbalanced RCAs. Volume control is through a digitally controlled, analogue resistor network. For this test, I will not be using the analogue bass management or tone controls that could affect signal presentation / quality. SNR for this device is rated at >110dB across the board for both RCA and XLR operation.

Denon AVR-3802:
A 7.1 channel "classic" from the decade when SACD and DVD-A were starting life and home theater lovers started seriously investing in discreet surround receivers with Dolby Digital and DTS (as opposed to the matrix surround of previous Dolby Pro Logic receivers). Analogue 7.1 channels input available. I bought this unit back in early 2002. Though not the highest end back then, it wasn't cheap (I think I paid almost $2000CAD). I don't remember the results of actual 3rd party testing but the rated amplifier power is 105W into 2 channels at 8ohms with 0.05%THD.

Analogue input SNR is rated at (only) ~86dB based on the table in the manual. For the sake of measuring the best possible audio output, all measurements will be performed through the CD input in "DIRECT" mode which bypasses all processing including tone controls and bass management. Measurement will be off the front (stereo) channels of the 7.1 "pre-out" analogue outputs.

ONKYO TX-NR1009:
My new receiver mentioned in the previous post. Capable of 9.2 channels processing with 7.1 analogue external inputs. Again, not the most expensive in the Onkyo line but certainly in the upper end of the previous generation released in mid-2011. Amplifier is capable of more power than the Denon with a rating of 135W into 2 channels at 8ohms with 0.08%THD. Sound & Vision measured it as 145W into 2 channels at 8ohms with 0.1% distortion. Line level SNR rated at 110dB.

I'll measure it through the analogue CD input in "Pure Audio" mode where all extraneous audio processing is turned off. Likewise, it's supposed to quiet the video circuitry and even the front LED indicators and display are turned off. I will measure from the front (stereo) channels of the 9.1 "pre-out" analogue output.

Firmware was updated to the latest version as of September 2013 - 1131-1399-0211-4108.

Setup:

 

First, I just want to discuss the general setup. The main thing I wanted to know was just how much resolution was maintained with the signal going through the preamp and look/listen to the results through a variety of levels standard across each device.

For the TEAC UD-501 DAC at 24/96 (which to me is the sweet spot) using the "SHARP" digital filter measured the following way:

HTPC (AMD A10 "Trinity") --> shielded USB (Belkin gold) --> TEAC UD-501 --> shielded RCA --> E-MU 0404USB --> shielded USB -->  Win8 laptop

we get these results:

The hope of course is that when we pass the above signal through the pre-amp, there will be minimal loss in resolution (noise level remains low around -113dB, and no change to frequency response to suggest "coloration" of the sound). In my previous TEAC measurements, I noted that the XLR output was too "hot" for the E-MU 0404USB without volume attenuation which drops the resolution. My guess would be that the noise level drops to less than -116dB. I will measure the XLR output from the Emotiva XSP-1 and see (it's the only device out of the 3 capable of balanced operation).
 
Using the digital oscilloscope, I found the following correlation between peak voltage output from each preamp device (accurate to <0.05V) and the volume control setting (using the TEAC RCA or XLR input of course):

Nice to see the volume control accuracy of each device - every halving of output peak voltage corresponds with a 6dB decline of the volume "knob". Notice that the Emotiva and Denon are using a relative system of volume control (dB below 0dBFS) and the ONKYO is set to an "absolute" measure between 0 to 100.

The setup incorporating the pre-amp in-line therefore looks like this:
HTPC (AMD A10 "Trinity") --> shielded USB (Belkin gold) --> TEAC UD-501 --> shielded RCA/XLR --> Pre-Amp device --> shielded RCA/XLR --> E-MU 0404USB --> shielded USB -->  Win8 laptop

Results:


I. Emotiva XSP-1 RCA:

Without further ado, here is what the Emotiva looks like with the unbalanced RCA setup:


Frequency Response

Dynamic Range
This looks really good. At 2V peak output, the dynamic range at 111dB is very close to the "ideal" (113dB directly from the TEAC). Notice a very small amount of roll-off in the high end when using the Emotiva. Also as expected, when the volume is reduced (2V --> 1V --> 500mV --> 250mV), the signal-to-noise ratio goes down and we see a concomitant reduction in the dynamic range and rise in noise level.

II. Emotiva XSP-1 XLR:

Let's have a look at the XSP-1 operating in a balanced configuration:

Frequency Response
Dynamic Range
Well folks, proof that if you want absolute fidelity, you really need to squeeze out those last few dB's down below 110dB with balanced XLR cables! Irrespective of whether you can hear it or not :-)!

Seriously, these are some fantastic measurements. As I said previously, unfortunately, I don't have direct measurements for the TEAC's XLR output. When passing the XLR output from the TEAC to the Emotiva pre-amp, the results at 2V peak volume coming out of the Emotiva slightly exceeds the direct RCA output from the TEAC DAC across the board from noise level to distortion levels to even lower stereo crosstalk.

The high frequency roll-off is less than with RCA. Notice just how clean the dynamic range graph looks as well through XLR cables. Fantastic.

III. Denon AVR-3802 RCA

Now we get into the AV receivers:

Frequency Response
Dynamic Range
Remember, I am measuring the Denon in "DIRECT" mode with all audio processing including bass management turned off (front stereo speakers set to "Large" for the sake of completeness). Not unexpectedly, these results are clearly a step down from the Emotive XSP-1. With a dynamic range of ~96dB at 2V, the Denon is capable of passing through 16-bit CD resolution but nothing more.

Roll-off above 20kHz is similar to the Emotiva's unbalanced mode but worse than the Emotiva below 100Hz with a -1dB bass roll-off at 20Hz.

IV. ONKYO TX-NR1009

Finally, let us have a look at what approximately a decade of advancement (between this and the Denon) in AV receiver technology can do:


Frequency Response
Dynamic Range
With the ONKYO in "Pure Audio" mode, no video processing at all, HDMI and video inputs all disconnected... Wow! That's very impressive IMO for a machine that's a "jack of all trades". In fact, these results are almost the same as the Emotiva XSP-1 functioning in unbalanced mode!

As excited as I am about those results above, a modern AV receiver is meant to process HDMI and be connected to a TV. This receiver has a HDMI "passthrough" which is essentially always in operation and for most people, it would not be left in "Pure Audio" mode with all the video gear disconnected. As such, look what happens when I connect my LG 55LW5600 TV (55" passive 3D, LED TV from 2011) to the ONKYO and repeated the measurements:

Frequency Response
Dynamic Range
Ugly, my friends... Clearly having the TV HDMI connected has injected very significant amount of noise in the system! Dynamic range has dropped to ~80dB across the board (equivalent to 13-bits). Notice a very strong 60Hz mains hum which is even showing up in the frequency response graph... What is happening here is that I'm seeing the effect of ground loops. There are ways to overcome this of course. For example, using a 3-to-2 prong adapter to disconnect the TEAC DAC from ground resulted in the following:

Frequency Response

Dynamic Range
About 10dB improvement just by doing this. For now, I'm not going to bother isolating the problem further (I'll be moving house in about a month!) but suffice it to say that in a receiver setup with complex component interconnections, be very careful of noise polluting the analogue output as demonstrated above. Ground loops are very common especially with TV systems where ethernet and coaxial cables are often connected to the TV/receiver creating a number of potential ground points beyond the individual device plug-ins to the wall.

Summary:

 

So there you have it. The Emotiva XSP-1 measures as a very capable pre-amp unit with excellent resolution especially when used in a balanced configuration. There was barely any loss through RCA and the XLR performance is beyond the E-MU 0404USB's measurement capabilities. Note that in all these tests, I'm just using generic "Radio Shack" type RCA connectors and the XLR cables are inexpensive Monoprice brand. No reason for spending money on expensive cables when these kinds of results can be obtained with standard decent interconnects.

In "Pure Audio" mode without the HDMI system connected, the ONKYO performed excellently. It bested the 12-year old Denon AVR-3802 handily and is essentially neck-and-neck with the Emotiva in an unbalanced configuration. However, beware of the potential noise pollution and ground loops once you plug in all sorts of things into these receivers (like your fancy big screen TV)!

A little while back, I spoke about how music sounded better through the Emotiva XSP-1 compared to using the Denon as pre-amp. These results are supportive of my subjective impressions (I'm showing >10dB dynamic range difference and bass roll-off differences between the two). As for the ONKYO, it does sound much like the Emotiva as a stand alone audio device. Since I will be listening primarily with the ONKYO as a HDMI DAC (either from the computer or through Blu-ray player), for me the digital audio performance is much more important than the results I show here from the analogue input.

Music this evening:
My kids enjoyed Les Misérables (the movie) and really love to listen to it in the car on the way to school... My favorite recording of this is the recent 25th anniversary UK Tour cast from 2010's performance called "Les Misérables Live!" - certainly much better singing than Hugh Jackman and Russell Crowe!

I've got another trip coming up at the beginning of November and then the house move by the end of November. I hope to put up some results of the ONKYO as HDMI DAC before I go. Until next time... Enjoy the tunes...


MEASUREMENTS: ONKYO TX-NR1009 as HDMI / SPDIF DAC... Are AV Receivers any good?

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The rat's nest.

Running separate components like multichannel processors/preamps to monoblock amplifiers are generally considered the ideal, "cost no object" approach to home theater. In the real world, cost and space are considerations and AV receivers become the "Jack-of-all trades" central device that most of us have in the home theater setup. But like the proverbial "Jack", it's useful to also consider the second part of that saying... Just how bad  is he also "the master of none"?

In the last installment, we looked at passing an analogue audio signal through the Onkyo and found that noise can be an issue. Today, I want to demonstrate the quality if we were to just use this device as a DAC - a look at the digital portion. Some natural questions arise - how well did the designers shield noise from getting in (especially in light of the high analogue noise measured previously)? Is the jitter through the use of HDMI "bad" (compared to TosLink and coaxial S/PDIF)? How does it compare to other stereo DACs?

Based on the Onkyo specs sheet, the TX-NR1009 uses the TI PCM1690 6-channel + PCM1789 2-channel DAC chips. Both are rated as 113dB SNR. These DAC chips are often found in consumer AV receivers and are lower spec'ed than most stand-alone DACs like the TEAC UD-501's PCM1795 with >120dB SNR. Of course, you cannot just look at this specification and judge the quality of a DAC. Much depends on the analogue circuitry around that DAC so the measured results are more useful than just looking at the components individually.

Setup:

As usual, for the sake of full disclosure and opportunity to repeat/verify, here is the setup for these measurements:

Win8 AMD A10 "Trinity" HTPC --> HDMI/TosLink/coaxial cable --> ONKYO TN-NR1009 front stereo "pre-out" --> shielded RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop
CM6631A device used for asynchronous USB --> coaxial / TosLink conversion duties.

HDMI driver: default AMD WASAPI. I used JRiver for playback.

Since I want to check the performance in a more "naturalistic" fashion, I made sure the TV was connected and on as well as my Blu-ray player (Panasonic BMP-TD220). Remember that in my previous post, plugging in the HDMI TV cable added significant noise to the analogue pass-through. All results were made with the Onkyo in "Pure Audio" mode to defeat any audio DSP/bass management.

HDMI cable: A decent looking 6' length capable of high speed HDMI (officially rated as HDMI 1.3 but fine with my HDMI 1.4 3D TV), brand named "ION" that I purchased for something like $20 about 2 years back at a local computer/supplies store.


TosLink and coaxial SPDIF cables I'll be using for comparison are the "Acoustic Research" branded 6' lengths I measured previously (see links for details).


Results:

As usual, I ran the output through my digital oscilloscope first to have a look:
Here is a 0dBFS 1kHz square wave sent through the HDMI and measured off the front stereo "pre-out" RCAs. Not bad - good square waveforms with excellent channel balance (sorry about the pixellation, usually screenshot looks better than that). With the receiver volume set to the "reference" of 85 (there is a little popup on the front screen when you hit 85 that correlates to the 0dB THX reference level) and in "Pure Audio" mode, the peak voltage is around 2.3V.

Here's a 24/44 impulse response:
Good linear phase impulse response, nothing fancy here. Absolute polarity also maintained by the Onkyo.

I. RightMark:
As usual, I used RightMark to look at the measured dynamic range, noise level, distortion levels; here's the summary for HDMI at the various bit depths and sample rates:

As you can see, I've also included for comparison the results at 24/96 for the Squeezebox Transporter and TEAC UD-501 (unbalanced outputs) - two of the best measuring DACs I've tested here with the same hardware/software.

Clearly the Onkyo is capable of hi-res with >16-bit dynamic range. With 24-bit data, it can do ~109dB dynamic range which equates to just over 18-bits! Not as good as the dedicated audio units like the Transporter or TEAC but pretty darn good for an AV receiver! This result is about equivalent to the AUNE X1 and ASUS Essence One using unbalanced RCA output - however, those DACs had better distortion numbers.

Some graphs to review from the 24/96 dataset:
Frequency Response
Noise Level

THD
The Onkyo rolls off a bit more in the high end, a little more noisy, and notably more harmonic distortion.

For fun, here's the spectrum off the Onkyo playing 24/192:
Yup, capable of 24/192 although the roll-off on the high end is obviously earlier than the TEAC UD-501 (Onkyo drops -3dB at 50kHz).

I was curious if the SPDIF (TosLink and coaxial) inputs measured just as well:

Yup. They all look pretty good. The graphs all look identical except for slightly more high end roll-off with the HDMI interface compared to the SPDIFs - not sure why.



With the dynamic range >16-bits, this test should be no problem for the Onkyo (HDMI input).

That looks very nice given that many very expensive DACs are not even capable of this degree of resolution! Again, this is an AV receiver! As I previously posted, even recently released DACs like the Wadia 121 Decoding Computer is incapable of this resolution.

III. J-Test for Jitter
As usual for my DAC tests, let's have a look at the Dunn J-Test spectra for both 16-bit and 24-bit signals. Here is the summary using the 3 digital inputs - HDMI, coaxial SPDIF, and TosLink SPDIF:



As you can see, the Onkyo is quite jittery in general whether HDMI or the other SPDIF interfaces. Although quite similar, I am somewhat surprised that the sidebands were more pronounced for the coaxial digital input! For comparison, here's the Transporter and TEAC:


Although the scale and dimensions are a little different, one can certainly appreciate just how jittery the Onkyo is compared to the others especially with the 24-bit signal. From this data, we see that the Onkyo itself has more jitter as a whole; specifically it's not any worse with the HDMI interface. We'll talk about jitter again in a little bit...

IV. Does sending a 5.1 channel signal degrade the measured performance?
I thought this would be interesting to check out. I left the RightMark test signal as the two front channels and added some AC/DC "Thunderstruck" into the center, rear, and LFE channels played back in JRiver as a multichannel FLAC through HDMI.


Beautiful ain't it?! The idea is to see if driving 6 channels (5.1) at the same time through the HDMI cable into the Onkyo's DAC will change the audio quality... For example, doing this might increase the noise floor, or perhaps worsen channel crosstalk since we've tripled the number of audio channels being processed.


Frequency Response

Noise Level
As you can see, there's very little difference whether 2 channels are playing or 6 channels. Great to see! Essentially no frequency response or crosstalk difference. However, there is a very small increase in noise level when playing multichannel... IMO audibly insignificant but measurable.

Summary:

Here you go folks! That's how a higher-medium end "modern" AV receiver measures as a stereo DAC. Of course, each model will be a bit different, but I suspect similar tiered receivers from Pioneer, Denon, Integra, Yamaha, H/K, Anthem, etc... should be comparable (won't know unless someone tests it out). Note that most magazines like Sound & Vision will measure receivers but usually in the context of power output and flatness of frequency response rather than on the accuracy of the digital-to-analogue conversion as I did here.

In some ways I am impressed and in other ways the results were as expected.

I was impressed by the low noise and very good dynamic range for example. To achieve almost 110dB in the audible spectrum is quite something especially considering the complexity of an AV receiver with all the potential electrical noise sources inside the box! The accuracy of the 16-bit -90.3dB waveform looks excellent; something which only the better stereo DACs or CD players would have been able to accurately reproduce a decade back. Likewise, the fact that the measurements remained excellent even with 6 channels being processed concurrently and only measuring about 1dB difference in the noise floor again demonstrates the engineering quality. Given the results I found previously with HDMI noise polluting the analogue input, I'm guessing that Onkyo put more attention in optimizing the digital side which makes sense since most people will be connecting digital inputs for multichannel sound.

As for the more "expected" results, let's talk about jitter...

A few years ago in 2009 this message came over the 'Net which I remember made quite an impression on me around how "bad" HDMI is as an audio interface (supposedly from Hi-Fi News & Record Review / Miller Research):
(I didn't notice it at the time, but that Denon AVR-3803A was a typo - the 3803 has no HDMI. It's actually the 3808.)

Later, a more comprehensive message appeared:

Hmmm, it looks like HDMI jitter can be cleaned out after all (eg. Arcam, Classe, Pioneer)! It's about the implementation, not necessarily the interface itself. If you read around these posts, one also finds that the jitter value and subjective sound quality do not necessarily correlate.

Let's think about the J-Test and what was found in measuring the Onkyo for a moment. The Dunn J-Test is a synthetic test of data jitter first published by the late Julian Dunn around 1994 which (in the 24-bit 48kHz version) superimposes an undithered LSB 250Hz square wave over a primary 12kHz -3.01dBFS sine wave which is of course an exact 1/4 of the sampling rate. This superimposition stimulates the effect of subtle timing inaccuracies (jitter) which can be demonstrated as accentuation of the sidebands measured in the spectral graphs.

Remember that this test is synthetic and stimulative. What you see measured is not something you're probably ever going to "hear" in real music! The noise floor is not going to be down to the last bit in 16-bit audio and essentially impossible with recorded 24-bit audio (unless it's purely computer synthesized music). Also, jitter is more pronounced in the higher frequencies (11kHz and 12kHz are used as the primary signals in the J-Test). Realize that the human hearing sensitivity is well on its way down by 5kHz (as can be seen by the Fletcher-Munson Curves). Furthermore, if we specifically look at the Onkyo's J-test spectrum, the most pronounced side bands are about -90dB below the primary signal. To make matter even less worrisome is that the tall sidebands are all +/-250Hz around the primary signal and the audibility would be masked even if one did have awesome auditory acuity at 11/12kHz and could hear a signal 90dB down! This is also why I feel adding up all those sideband peaks and calling it a number (whatever picosecond or nanosecond) is really not all that useful when it comes to audibility.

What I'm trying to say is this... Tests like the J-test can demonstrate that jitter is a real phenomenon. Engineers should pay attention to it when designing hi-fi equipment. A discerning audiophile should be aware of it and if able to, can measure it themselves and decide if the engineer did a good enough job. However, IMO, to say that jitter is somehow audible at these kinds of levels I think would be impossible. In fact, unless the jitter were ridiculously high (like Track 26 from Stereophile Test CD 2, where an insanely simulated 10ns sideband is inserted +/-4KHz around the 11kHz primary - again totally synthetic), the concept of jitter significantly deteriorating sound quality I believe is utter nonsense in the real world. That some companies would even consider using jitter as a reason for putative significant audible differences between passive "components" like cables is just not credible!

I had a listen to the Onkyo's output over the last few nights with some familiar music - Ella Fitzgerald "Sings The George & Ira Gershwin Song Book", Grateful Dead "American Beauty", and Keb' Mo' "Just Like You". Also had a listen to Sting's new album "The Last Ship". They all sounded nicely rendered as they should with a good DAC. Great details with my older Paradigm Reference Studio 80 v2's which will be my rear speakers in the new sound room. The wife and I both enjoyed Sting's "The Night The Pugilist Learned How To Dance" - cute.

So, even though the AV receiver might be the "Jack-of-all-trades", at least in this specific instance with the Onkyo TX-NR1009 as an HDMI DAC, he might not be a "master" at it, but I'd say he's a pretty decent tradesman :-).

Well, unless I dig up something else to report, I'm likely "going dark" for the next month as I head off overseas for some work and then the big move to the new home. I'll be sure to post some pictures in time... Enjoy the music everyone!

GUEST REVIEW & MEASUREMENTS: The Quantum HDMI Squeezer + ULTRA Cable: A look at HDMI cables.

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By Keaton I. Goulden-Eyre III, Esq.

With Archimago overseas, he implored that I take a few moments to contribute to this most obscene of blogs (sorry dear readers, "objective" and audio do not mix in my worldview based on experience, wisdom, and my ears). Recall that many months ago, I brought you the review of Dr. Frank's "Best-Coaxial-Digital" SPDIF cable. I remain steadfast in my opinion of that fantastic interconnect!

A reminder - the introductory price is still available until December 31st! At $4999.99, it is a steal.

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The cellular phone rang - "How inconvenient!"

That was my initial thought the evening I heard about the cable being reviewed today. It happened in August as I was at my usual Las Vegas soirée with associates enjoying some Château Pétrus on my way to a Wolfgang Puck restaurant with a tender morsel of imported Wagyu in mind.

On the other end was Alfred Fitzgerald, LL.B. A member of my exclusive gentleman's club back home who could not wait to discuss an amazing audiophile find. Having had many deep conversations around our shared passion for audio reproduction, he knew that I would find his news intoxicating. He was correct.

It so happened that recently he was representing the interests of a client; Dr. Joseppi Maltzarelli, in the acquisition of a 30m yacht off the Florida Keys. He discovered that Dr. Maltzarelli was in fact a physicist who interned at CERN's Large Hadron Collider in 2005. His ground breaking theories on the vibrational qualities of quantum superstrings in the terahertz range drew applause but also envy from colleagues such that he decided to leave the "mainstream" physics community to become the chief scientific officer of a startup to leverage his theories and experience. The company: QuantaVibes Inc. based in Dushanbe, Tajikistan, aptly located just across the street from theGurminj Musical Instrument Museum showcasing the history of stringed instruments.

Although the phone conversation was brief, I could not enjoy the succulent Kobe that night - lost in thought as I contemplated the potential of what I heard! I just had to call Dr. Maltzarelli upon my return home. I was almost unable to enjoy Céline Dion that evening!

After many E-mails and calls to QuantaVibes, I was finally able to track down Dr. Maltzarelli via satellite phone located somewhere off the coast of French Guyana in his yacht. It was a wonderful discussion; here is a good portion of our conversation:

ME: Good afternoon Dr. Maltzarelli. Thank you for taking my call.

JM: Absolutely Keaton, any friend of Mr. Fitzgerald's is a friend of mine. (Both laugh in approval.)

ME: I am curious my good fellow, whatever are you doing out at sea?

JM: The South Atlantic is beautiful this time of the year, my friend! I'm planning to sail down to Brazil and into Uruguay by the spring time where I will go inland to research the acoustic resonances of spiritual earthenware of the Amazonian peoples. Just a well deserved vacation after years of programming my supercomputer to calculate some Super Large Numbers involved in Superstring equations. The company is almost ready to ramp up production on what we believe will be the most significant HDMI audio upgrade in this generation - if not the most significant audio upgrade of all time!

ME: Amazing Joseppi. I'm perplexed however, how did a physicist of your calibre ever get involved in audio in the first place?

JM: You see, this is what happened. In 2005 when I was at CERN, I discovered that free-electron lasers were capable of inducing terahertz vibrations in the Superstring subequations as expressed using the derivative of the semiconductor Bloch equations. This caused quite a stir in the community because it meant that resonance effects created by this perturbation in high order space-time could fold down into our 3 spatial and single temporal dimensions! My colleagues were not able to handle this truth. They started a smear campaign decrying my theories and even went as far as to label me as mentally unstable!

ME: Terrible! Such close mindedness - they probably still believe the earth is flat! Was this why you left CERN?

JM: Yes, amico. I could not tolerate this final insult and left to pursue other avenues to realize the profound implications of this research. As to the second part of your question; audio was a natural fit for these equations. Like the strings of a violin, the Superstrings resonate in a natural harmonic. It just so happens that these harmonics precisely overlap with the audio spectrum when free-electron laser spectroscopy is activated at odd harmonic multiples. As a result, we can precisely tame the stray frequencies and decouple the thermionic energy flux passed across various equipment. My research pin pointed to the HDMI interface as one which could use this "taming" effect as a first foray into audio reproduction for the company.

ME: So, is it something to do with the HDMI interface's complexity?

JM: Precisely Keaton. You see, HDMI transports "bits" like how the Transporter in Star Trek transports matter. HDMI communicates using TMDS which sends those bits and nibbles with no respect for timing or integrity. This just shreds the musical information apart and artificially reproduces it at the other end! No wonder people experience horrific digititis, headaches, gout, sarcoidosis, gastrointestinal problems, and other forms of neurasthenia with this wretched interface for music. Is it any wonder how jittery the HDMI interface is?

ME: Impressive, doctor. So what is this product soon to be released to combat the problem?

JM: Soon, we will be releasing the QuantaVibe Quantum HDMI Squeezer and accompanying QuantaVibe ULTRA HDMI Cable. They should be purchased as a pair for synergistic effect. The Squeezer consists of an adapter for regular HDMI to micro-HDMI because supercomputer simulations demonstrated that the smaller size of the micro-HDMI interface precisely corresponded to the wavelength of these Superstring terahertz vibrations. The increased density of electron flow through the micro-HDMI connector accentuates the resonant-transduction effect by 323%. We've treated this device with a patent pending ultramicrochip which precisely aligns the resonances. Unfortunately this is a trade secret so I cannot elaborate any further.

ME: And how about the ULTRA Cable?

JM: We understand if an audiophile wishes to use a standard size HDMI-A connector or cannot modify their system to accept the micro-HDMI. It is more convenient but no matter how I load my equations for the terahertz wavelength, it is still a compromise due to the size of that connector. Nonetheless, we will be selling separately our ULTRA Cable which has some of the technologies incorporated in the microweave of the insulator. Again, I cannot divulge any further information on the technology itself lest I get in trouble with the CEO of the company. (Both laugh.)

ME: You know Joseppi, many insane "objectivists" will be very critical of these worthy products. What do you have to say to potential critics?

JM: Keaton, my friend, there will always be "haters" in this world. I faced many back at CERN under the guise of "peer review" and still get many thumbs down with my Facebook posts and criticism with interviews like this one probably. I do not expect everyone to appreciate the benefits. In our extensive testing in the lab, only those with excellent hearing, trained ears, truly high-end equipment, and impeccable taste in music can fully enjoy what we are about to produce. Furthermore, we are so convinced that the discriminating audiophile will love this product that we will be offering a 35-day guaranteed satisfaction or full refund! Absolutely no risk! I do not believe anyone can beat that.

ME: I have never heard of this kind of offer - 35 days! Now how about sending a set to me for a review?

JM: Absolutely, sir. I will have my people contact yours. I apologize Keaton, I have to go now, the fidanzata is calling... Never let the fidanzata wait...

ME: I absolutely understand. I look forward to the review sample and our next opportunity to converse. Perhaps at an audio show? I hope you find some hidden truths in the spiritual earthenware of the indigenous Amazonians.

With that, and good to his word, a package arrived from Tajikistan two weeks later. Neatly secured in its own black silk pouch I found this set of adapter and HDMI interconnect:

The workmanship was excellent. Black which matches my custom-designed HDMI DAC (connectors personally soldered by Nodko-san in Japan) with fabulously gold plated connectors. Directionality was clearly marked on the cable (not shown). I was informed by an associate at QuantaVibes that these are prototypes and the production units will feature gold embossed lettering on aerospace-grade titanium in place of the white label shown above.

The Quantum Squeezer is a mere 6cm (2.5") in length but tucked within it the full package of Superstring optimizations. At perhaps 15 gm in weight it was ethereally light - befitting of the level of technology! The ULTRA Cable is 12' in length and should be long enough for essentially any connection between your source and the HDMI DAC. This cable was optimized for music so do not expect it to carry nonsense HDMI 1.4 extensions like 3D or even 1080P to some sources*. Wow! Mind boggling how the potency of these optimizations were capable of limiting video transmission in the service of audio.

"Ultimate Smooth"
I immediately connected up the Quantum Squeezer and ULTRA Cable to the HDMI input of my UltraBook computer and custom DAC for a listen. (Remember, the Quantum Squeezer only works with the micro-HDMI port common on newer portable computers like laptops/ultrabooks.)

I don't know how Dr. Maltzarelli did it, but he did! I swear, the Herbert von Karajan & Berliner Philharmoniker Beethoven Symphonies played in my soundroom from my 16-driver 863 pound custom speakers with GIA FL grade diamond tweeters driven by special edition Nodko 8-Watt SET tube amps with a sparkle and clarity I had not thought possible. The strings were smooth like a well aged Highland single malt scotch whiskey or the hum of my newly acquired Jaguar F-Type V8 S. The timbre of each instrument resonated with a "note" beyond the vocabulary of the best Wine Spectator writer. This was the power of Superstring audio optimization!

The beautiful multi-layered vocals from Stephen Layton & Britten Sinfonia's version of Chorus: For Unto Us A Child Is Born (off "Handel's Messiah", 2009) almost dislodged me from my seat with the immensity of the recording venue's soundstage (St. John's, Smith Square, London). I had never heard the numerous voices with such definition. I could make out the fact that the tenor in the second row secretly picked his nostril at 1:23 into the track. Replacing the QuantaVibes cables with my AudioSearch Whiskey HDMIs ($300/3') demonstrated just how superior the QuantaVibes were and stepping down again to generic HDMIs ($20) resulted in either a collapse or dissolution of soundstage, leaving the voices decapitated, floating in space in one instance and the next congealed in an incomprehensible mess as if lying supine in a morgue. The joy was gone, the textures made bland, soulless. It was so obvious that anyone who could not tell the difference must be auditorily blind.

The same effect could be found with more pedestrian music. Consider the sitar on the Beatle's Norweigian Woods (off the 24-bit remaster of course). On the vast majority of HDMI cables (including very expensive ones I might add), they sound shrill, overly trebly, and ethnic. Through the Quantum Squeezer and ULTRA Cable, this instrument played with its full glory demonstrating George Harrison's connection with the numinous (perhaps aided by various hallucinogens?), altogether natural, at One, familiar. This level of sonic reproduction is priceless!

Finally, my luscious wife Candy wanted to participate in the audition.We cued up her favorite track from the Spice Girls - If You Wanna Have Some Fun. The soundstage exploded beyond the walls side-to-side, front-to-back; and the vocalization from the Girls were lined up beside each other - you could even discern the relative heights of these women! Candy squealed in delight exclaiming "I ain't heard it go so low before, Big Daddy!" Indeed, we had fun.

Readers, let me be perfectly clear about this. Forget all you have heard about "holographic" sound from inferior equipment. These cables offer HOLODECK sound. Miles Davis' spit could be heard and visualized dripping off his trumpet, Coltrane's sax keys rattled before my eyes, Elvis' hips gyrated in concert with his live performances, and Michael Jackson's lewd gestures beckoned beyond the grave off Thriller! Dare I say, this is the first time I have experienced digital sound even begin to achieve parity with my vinyl collection. Such was the presence. You know you want this.

Now, as per my agreement with Archimago, measurements are a pre-requisite for reviews of such gear on his blog (again, absurd). I lent him the QuantaVibes HDMI Squeezer and ULTRA Cable for a couple nights just before he went off on the plane. I'll be back in a moment after this unnecessary interlude...

Objective Analysis:

Okay, as Keaton said, I had the chance to measure his review cable with a couple other HDMI cables using the following setup just before I leave:

AMD A10-5800K HTPC with HDMI-A connector or ASUS Taichi with HDMI-D (micro-HDMI) --> Test HDMI cable --> Onkyo TX-NR1009 --> shielded RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop for analysis

Cables tested:
1. HDMI "high speed" cable, ION branded, 6' long, used in my previous Onkyo DAC measurements. No problem with HDMI 1.4 functionality like 3D to HDTV. Cost - about $20.


2. Fancy 4m (~13') Energy branded HDMI with all the check boxes ticked for HDMI 1.4a. It's touted as the "Connoisseur" Series which I'm sure Keaton would approve of. High speed to "13.8Gbps" specified, ARC, 3D, ethernet, 4K... Even has a tag as seen in the picture with "Confirmation of HDMI ATC Testing" - ATC in this case means "Authorized Test Center" and for this cable, the center was "Dat Tran". Nice metal connectors and general cable build quality.

At $50, this is probably the most expensive cable I have and will be used between the Onkyo receiver and LG 3DTV in my new setup. Note that the upcoming HDMI 2.0 standard specifies data transfer rates up to 18Gbps but is backward compatible with high-speed cables so I hope this cable will do the job in the years ahead.


3. The review QuantaVibes Quantum HDMI Squeezer + ULTRA HDMI Cable. The HDMI Squeezer looks like a standard HDMI-A female to HDMI-D male converter capable of a 90-degree rotation. Although said to be "heavily modified", I do believe similar adapters can be found at the local Radio Shack :-).

The ULTRA Cable is "standard speed" - tested to be OK with 1080P with my TV and the ASUS Taichi ultrabook, but NOT fast enough for 3D video off my Panasonic BMP-TD220 Blu-ray player.

Starting with the usual RightMark measurements, here's the summary (all done at 24/96, HDMI WASAPI driver):

Frequency Response

Noise level
No difference folks! Frequency response, noise level, distortion levels appear indifferent to the HDMI cables used.
 
Let's look at jitter with the usual Dunn J-Test:



Hmmm, what's this? The QuantaVibes spectrum is more jittery - especially noticeable with the 24-bit condition. However, notice what's happening here. Both the ION and Energy cables are being measured off the standard HDMI connector whereas the QuantaVibes is off the ASUS Taichi laptop's micro-HDMI port. What happens if we use the "HDMI Squeezer" converter but with the fancy Energy HDMI cable instead?


Voilà, the jitter spectrum now looks like the one above with the QuantaVibes ULTRA HDMI cable. Basically what is demonstrated here is exactly what I saw with the TosLink, USB, and coaxial digital interfaces. The jitter spectrum is a function of the sending and receiving device. The cable itself does not change the pattern; in this case, the little ASUS Taichi ultrabook tends to show more jitter than the HTPC AMD A10-based computer. Whatever HDMI cable is used does not change the jitter pattern (although I suppose one could wonder whether the HDMI-A to micro-HDMI adapter has an effect; not likely).

Bottom line from the objective side:No evidence that HDMI cables make any difference to standard measures of frequency response, distortion, noise floor/dynamic range of the DAC (Onkyo TX-NR1009 in this case). Jitter remains a function of the active devices, not a property that varies with the passive cables themselves (at least within the reasonable lengths tested up to ~13 feet). I'm happy to be proven wrong if anyone else has good data especially with less jittery DACs than this receiver.

I am therefore at a loss as to Keaton's enthusiasm around this product.

Back to Keaton for his conclusions.
 

Keaton's Konclusions:

Bollocks, more squiggles from Archimago... Yet again, measurements remain insensitive and unable to achieve the resolution of my 73-year young experienced ears. Hence useless and invalid for audio evaluation. We all know that everything matters, even more so digital cables because there isn't such thing as digital according to these enlightened gentlemen. HDMI is of course the worst of all the digital connections for audio (some other enlightened gentlemen at Audio Asylum agree) which makes it so much more important that we spend more money on ensuring perfect digital transmission.

As a side note, I connected the ULTRA HDMI Cable by itself to my Blu-ray player and 85" 4K TV. I swear, the image was more stable, colors brighter, and the actors moved so smoothly and with such poise that B-movies seemed Oscar worthy. Likewise, the audio-video synchronization was even better with these cables that I wondered how I managed to watch movies without them! Here again, the power of jitter-free sympathetic Superstring resonance at work. Indeed, I will be ordering a separate set of ULTRA's just for the Blu-ray player when the final product is released. Nonetheless, I feel that without the HDMI Squeezer, the synergism just wasn't there. The sound didn't reach as deep, the trebles didn't quite touch the heavens; without doubt, you need the full set!

You likely are aware that so far I have not said how much these high-tech devices will be sold for. I was told the company is still perfecting the quality control due to the precise manufacturing standards and complexity of the patent pending process. Expectations are that both the HDMI Squeezer and ULTRA Cable will be priced as a set at the $3000 mark. By itself, the ULTRA Cable will be around $2000. Mere pocket-change for this level of sonic/video revelation - I bet your power cords costs about just as much and they only have 3 individual wire lengths inside at most, and require less precise shielding! This is very comparable to other high-end HDMI cables such as these or these or these especially given the improvements on the quantum scale!


With this premier product from QuantaVibes, I am confident that we will 'hear' more from this up-and-coming newcomer to the high fidelity audio scene. I have a strong feeling that Dr. Maltzarelli's research into Amazonian earthenware will yield many revelations into audio resonances for upgrading the sound room. It was with supreme regret that I had to return the review cable back to QuantaVibes after 3 weeks of audio bliss for fear of industrial espionage. Currently awaiting their formal release with bated breath and ample liquidity in hand.

Until next time; Magico wishes and Burmester dreams.

-----------------------------

* This cable is rated as "Standard HDMI".

Ed: And so ends the digital audio cable measurement quadrilogy. That is, until yet another digital audio interface shows up with fancy cable claims... Enjoy the music till then :-).

Getting There... (Early HT Room Setup)

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Back from my overseas business trip late last week. It's going to be really busy since I'll be moving to the new house in 1 week. Massive amounts to pack up!

Nonetheless, I didn't want the movers to be involved with the audio system so over the weekend I moved all the components and put together the stereo setup in the home theater room to just have a little "taste". Here she is... (Unfortunately the image is a bit grainy. Resorted to the old Nikon D70 as my D800 developed some autofocus issues and needed repairs.)




Hmmm. Looks like I need to straightening the power/cable/ethernet outlet at the back there...

Room size is decent: 20' x 15' x 8'. The TV is a 55" LG 55LW5600 mounted on a strong wall mount - I might go for a 70" in the future. Components:

- SONY SCD-CE775 SACD/CDP I bought back in 2001
- Emotiva XSP-1 pre-amp
- 2x Emotiva XPA-1L monoblocks connected to XSP-1 with Monoprice XLRs - 35W Class-A bias, max 250W Class A/B
- Paradigm Signature SUB 1 subwoofer crossed over at 50Hz
- Cables: 4' 12G OFC Monoprice zip cord speaker cables, Radio Shack 3' RCA from CDP to preamp, stock power cables

I still don't have the sofa sectional in the room and room treatments, nor have I set up the SUB 1's programmable subwoofer room correction yet with PBK-1. Despite the bare room reflections, it sounds pretty decent still... Played my old Kind Of Blue SACD, Diana Krall's When I Look In Your Eyes, and Beck's Sea Change tonight. Really liking the subwoofer's punch on the Beck SACD. I'm an advocate for a good subwoofer... Good frequency response down to 20Hz is essential for hi-fidelity IMO.

Chat later... More boxes to pack tonight :-(.

MEASUREMENTS: Do lossless compressed audio formats all sound the same?

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The year is 2013.

Digital audio has been around for a long time. The CD 16/44 PCM format has been the de facto standard of audio delivery for 3 decades now. For the last decade at least, many of us have been involved in computer audio of one form or another. Personally I started seriously archiving all my CD's with bit-perfect rips since 2004 and conversion of all my PCM audio to FLAC by 2005/2006.

In all these years, I do not believe I have ever felt that playback of a compressed lossless format like FLAC compromised sound quality. Yet, if you look around the Internet at the various audiophile forums, you hear from all kinds of folks how uncompressed formats like WAV and AIFF "sound better" than the lossless compressed formats like FLAC, Apple Lossless (ALAC), WavPack (WV), and Monkey's Audio (APE).

Let's have a look...

Setup:

MacBook Pro (Decibel player) --> shielded USB --> TEAC UD-501 DAC --> shielded 6' RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

MacBook Pro is the 17" early-2008 model previously described. Nothing fancy, and in fact relatively "old" 2.6GHz Core 2 Duo processor. Running OS X Mountain Lion with no OS tweak for audio. Decibel version 1.2.9 (haven't upgraded to latest version yet). For Decibel I did not even turn on the "load file to memory" option so the lossless decompression is happening real-time.

Win8 laptop is the Acer Aspire 5552 which has been my measurement "work horse". Again, nothing fancy, just 2.2GHz AMD Phenom X4 processor to grab data from the E-MU 0404USB and process the data through DiffMaker.

Procedure:

I encoded the DMAC Test using dBPowerAmp 14.3 from FLAC (which I used to standardize the test results) into the various formats supported natively: WAV, AIFF, ALAC. I downloaded the binaries for APE (v.4.11) and WavPack (v. 4.60.1 Windows) from the official web sites respectively. I used the highest compression level available for each - level 8 for FLAC, -hh "very high quality" for WavPack, "Insane" for APE.

All files were transferred to the Mac and played back off the machine's 240GB SSD drive. I ran 3 iterations with each file format to account for some inter-test variability. DiffMaker comparison was made between my "standard" FLAC recording and each of the test recordings.

Results:


All the lossless formats scored within a narrow range. Correlated null depths across the board were in the 80-90dB range for the lossless formats. As expected, the lossy formats (MP3 and AAC) did not score as well. Also as expected, AAC 192kbps showed less variance (spectrally more accurate) than the equivalent MP3 encoded at 192kbps - AAC is newer and clearly better at lower bit rates.

Conclusion:

A couple observations...

Firstly, notice the greater variability in numerical results for the lossless formats (but remaining in the 80-90dB reference range). Remember that the correlation scale is measured in dB's - it's logarithmic. With "bit-perfect" measured correlations up around 90dB's, sensitivity is very high and it doesn't take much difference to alter the measured value. Measurements with results lower down like in the 60's and 50's tend to show less inter-test variability.

Secondly, when I listen to the "difference" WAV file produced by DiffMaker of 80-90dB correlated null depth, I need to turn up the headphone volume on the TEAC (listening with Sennheiser HD800) to maximum where it still sounds soft. With normal audio, this would be uncomfortably loud. Sonic differences therefore would be orders of magnitude softer than the normal music itself.

Bottom line. The measured variance from the TEAC DAC analogue output between lossless file formats decoded using an older Core 2 Duo computer without decoding into RAM first is extremely low - basically, there's no difference in the sound.

Do lossless compressed formats all sound the same? YES, they should, and in this test, they do.

Based on what I'm hearing and measuring, it's obviously not hard to get good bit-perfect sound. If a piece of equipment is producing audibly different output from say WAV vs. FLAC (that is, assuming the difference isn't cognitive/perceptual bias), then I think there's something wrong with the setup since this was not the intent of the creators of lossless compression. Either the settings are wrong (eg. transcoding to lossy format, ReplayGain tags being applied, or DSP turned on) or there's something 'broken' in the decoding process (eg. CPU too slow, data transfer speed issue, or poor software unable to keep up with the relatively low processing demands). This is a problem and diagnostics should be run to determine how to fix it.

As usual, please feel free to drop me a note or link to good evidence if you run across any information contrary to these test results and opinion.

Enjoy the music...


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Addendum (those interested in spectral plots of the difference between FLAC reference and test file):

FLAC / APE / WV / ALAC / WAV / AIFF all look somewhat like this - not much to see. Note: There's always a little bit of noise in measuring the analogue output plus limitations of the E-MU ADC.:


This is what lossy looks like in comparison - quite striking how much can be "reconstructed" and still sound good!
MP3 320kbps:

AAC 320kbps:

MP3 192kbps:

AAC 192kbps:

MUSINGS: A Look At The Sound Room...

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It's about 2 weeks now since I moved into the new home. As expected, it was stressful; with kids in tow, it's not like the bachelor days with just one's personal belonging in the trunk of my old hatchback!

Well, for the most part, the move hasn't been horrible. And the exciting thing as I've already previewed is that I now have a good sized "man-cave" for my A/V "needs" :-). Without further ado, let me show you the setup so far:
Head on view of the main system - Transporter playing.
Angled from the side - note the SUB1 just lateral to the left front speaker. The black box closest to edge of the sofa is the computer (Fractal Design Define XL case - nice and quiet).
"Super deformed" wide angle view of the room... Rear Paradigm Studio 80s visible.
As a MUSINGS post, I'm just going to spend a few words on thoughts around building the setup. Since the start of this blog and explored briefly in this early post, I do believe that judging the quality of an audio component can be accomplished objectively; that is, there is a technically "good" or "bad" way to know whether components live up to engineered goals. As I mused in that post, the gold standard for me is not so much to reproduce the "concert experience" as some might desire, but rather the ability of the equipment to reproduce exactly what's on the CD/DVD/DVD-A/SACD/Blu-Ray - that is my definition of high-fidelity. If the CD has an ability to transport me to a faraway concert hall, then I want my equipment to be able to reproduce that encoded sensory stimulus which leads to (hopefully) my ability to experience the same. Of course, not all CD/DVD/DVD-A/SACD/Blu-Ray's can do this! The medium itself must be able to encode quality to the extent that the experience is possible and the experience itself is provided by the artist, recording engineer, mixer, mastering engineer, producer, etc... who have put their expertise and knowledge into the recording. On my (consumer) end, I'm just looking for a good enough combination of components that can take that encoded sensory experience and provide it accurately; nothing more. I do not personally believe in aiming for a "euphonic" setup where the components can make all albums sound "sweet". I'm interested in just an honest presentation of what is on the disk; if I want to add euphonia, I will happily do it myself such as the PCM-to-DSD upsampling process or re-EQ with my Behringer DEQ2496.

As with anything in life, unless I were a billionaire, there are practical limitations on how much I am willing and able to spend on a sound system. I'm happy to sink money into the pursuit but it's only one of many interests! To spend as little as reasonably possible to achieve the best sound quality (and within decent aesthetic parameters) is a virtue I strive for. My experience has been that for electronics gear, there really is very little correlation between price and the (objective) sound quality it buys. For example, there's generally very little if any audible difference between a $500 DAC compared to a $2000 DAC if they measure the same; speakers, room acoustics, amps will easily trump the sonic differences. I can enjoy the inexpensive AUNE X1 (<$200) just as much as the more expensive TEAC UD-501 (~$800) even though I know the TEAC measures significantly better. In fact, I prefer the AUNE's more powerful headphone amp when I'm listening with the AKG Q701 headphones. However, the TEAC offers native DSD playback which is the niche it fills in my system. Likewise, the Transporter is my hi-res ethernet streamer, and the ASUS Essence One lives on my desktop for computer listening on account of its separate headphone/speaker controls and beefy headphone power. Where cost does seem to correlate better IMO is with the transducer devices - headphones and speakers. For these components, I'm quite happy to sink $$$ down! For fun, here's approximately how I've allocated out the cost of the audio system (minus HTPC which is more powerful than I really need for audio purposes). Note that I did not sit down to calculate this out before hand, it just organically came to be:

Speakers (fronts, rears, center, sub): 77.5%
Digital sources (including Behringer DEQ2496 processor, Panasonic Blu-Ray): 10%
Amplifiers (including Onkyo receiver): 10.9%
Cables + Belkin PF60 power console: 2%

Alright, I'm pretty happy with those numbers - I think they reflect reasonably well my priorities. Of course, the cost that truly trumps everything is the cost of real estate in Vancouver! So, let's run over the components I have set up in the room and share a few thoughts... As usual, since this is a MUSINGS post, it's mainly an experiential discussion with opinions thrown in.

I. First, let's talk about the 2-channel signal path:

Most of my music is in stereo. Therefore good 2-channel reproduction is most important. Enough said.

Time and again, measurements of the ASUS Essence One, Transporter, and TEAC UD-501 have demonstrated the superiority of balanced cabling. Whether anyone can hear the difference of course is another issue. Balanced operation was the reason for the choice of the Emotive XSP-1 preamp as the heart of the 2-channel system. To maintain the balanced topology, I got a couple of Emotiva XPA-1L monoblock amplifiers - good price and with the option to switch over to 35W Class A bias if I want.

Squeezebox Transporter music server chain:
Win8 HTPC --> ethernet --> Transporter --> Emotiva XSP-1 preamp (crossover at 60Hz to feed SUB1 subwoofer) --> Emotiva XPA-1L monoblock --> Paradigm Signature S8 v.3

Computer audio PCM/DSD chain:
Win8 HTPC --> Belkin gold USB --> TEAC UD-501 DAC --> Emotiva XSP-1 preamp (crossover at 60Hz to feed SUB1 subwoofer) --> Emotiva XPA-1L monoblock --> Paradigm Signature S8 v.3

Although just "fast ethernet" (100Mbps) is all that's required for the Transporter, the house is wired for gigabit and I've used generic Cat-6 cables for the Transporter to the gigabit switch. I'm quite pleased that I can easily transfer >100MB/s between machines around the home. All balanced audio cables were inexpensive (but good build) Monoprice Premier XLR's from 3-6' in length. Over the months, I've used Monoprice cables to measure balanced output from my DACs and the results have been excellent, not unexpectedly.

In the same vein, speaker cables are Monoprice 12-guage "Enhanced Loud" (LOL!) OFC. Monoblock to front speakers only 4', center channel 6', rears at most 25'; cut to minimum lengths required. I bought a 100' spool for $30 and still have some left.

II. Multichannel signal path:

I love multichannel music! The realism achievable can be amazing and IMO anything that enhances the creative potential of artists can't be a bad thing. Remember that historically multichannel speaker configurations were being explored along side 2-channel stereo. 3-channel stereophonic sound was demonstrated by Bell back in 1933 and the right-center-left "3.0" arrangement was used in some of the earliest "Fantasound" systems for Disney's Fantasia when released back in 1940. Having a center speaker in a theater setting allows the anchoring of front-and-center sound which improves the imaging for those not sitting precisely in the "sweet spot". For music, likewise it helps especially for solo/vocal tracks. For example, the Analogue Productions' Nat "King" Cole SACDs like The Very Thought Of You presented in 3.0 sounds phenomenal with this arrangement with Nat sounding like he's right in front of you crooning.

More than 10 years ago, I built a discreet multichannel system based on my old Denon AVR-3802 receiver. However, I had to give up the 5.1 setup when my kids came 8 years back to make room. After many years in "pure stereo" wilderness, I'm glad to finally be back with a full 5.1 setup again! Here's how it's hooked up:

Win8 HTPC / Panasonic Blu-Ray --> Energy HDMI --> Onkyo TX-NR1009 (amplifies rears and center, up to 145Wpc 2-channel measured) --> unbalanced RCA --> Emotiva XSP-1 preamp (HT Bypass Mode with channel to SUB1) --> Emotiva XPA-1L monoblock --> Paradigm Signature S8 fronts

Center speaker = Paradigm Signature C3
Rear speakers = Paradigm Studio 80 v.2 (tonal balance complements the Signatures reasonably well)

As you can see, my rears are full range towers.  I'm aiming for speaker layout angles approximating the ITU-R BS.775-3 (08/2012) recommendation at the "sweet spot" position:


I suppose I have room for a full 7.1 surround setup with 2 extra speakers to either side using the Onkyo receiver... Another SUB1 for 5.2 or 7.2 would give insane bass! One of many projects for the future, I suppose. I am unaware of any music I want available in 7.1 at this time and I suspect a TV upgrade to something like 80" would be more likely.

III. Challenges...

In a moderately complex setup, it's not surprising to find some challenges along the way. The main thing I found was that for multichannel, the Home Theater Bypass setting on the XSP-1 was very sensitive to noise. I had to move the PC to the left side about a foot from the subwoofer and ~5' from the XSP-1 to get rid of RF noise picked up by the preamp. Also, I had to use a "cheater plug" for the LG 55" HDTV mounted against the wall to remove ground loop noise. This was the relatively easy stuff!

The most difficult noise issue I'm still dealing with now is the USB interface to the TEAC UD-501. If I have the USB cable connected, there's a high pitched whine emanating in HT Bypass mode. This does not appear to be a component ground loop issue but rather noise from the PC through the USB interface polluting the analogue pass-thru. This actually does not affect stereo playback from the TEAC, just when I'm in multi-channel mode with the XSP-1 passing through the front stereo and subwoofer channels. It's not an issue with the RCA cables since more expensive AudioQuest and Tributaries RCA cables make no difference compared to inexpensive Radio Shacks whether 3' or 6'. The simple solution for now is unplugging the USB cable to the TEAC DAC when I'm listening to multichannel. Trying other USB ports and hubs have so far not helped. I'll have to look at other options like the FireStone GreenKey "USB Isolator" or some other way to achieve galvanic isolation but maintain high-speed USB 2.0 for DSD and hi-res PCM playback.

Looking ahead, I still have to try out some frequency response measurements and subwoofer room correction with the Paradigm PBK-1 ("Perfect Bass Kit") I bought (only ~$120). I'm inspired by Mitch's experiments with Acourate so may look into that too... The room is still bare and resonant so things should also improve when the rug comes and in time, perhaps some acoustic paneling and bass traps. Not to mention some clean up and better cable management!

As is, subjectively the system sounds good despite the lack of room treatments... Of course, I am a little biased :-). The Signature S8 v.3's are the current top-of-the-line Paradigm floor standers. Good to see some positive recent reviews like this one from TONEAudio. Some might consider them too "clinical" but that's fine with me since surgical accuracy is what I'm after. A large company like Paradigm can leverage the economy of scale to maintain costs and has access to research facilities which IMO is important. The beryllium tweeters sound sweet and very realistic. The other night my wife jumped when she heard the sound of the glass shattering on Michael Jackson's Jam (surely a sign of high fidelity!). So far I've also been quite impressed with the SUB1 subwoofer. I'm easily measuring excellent levels around 20Hz. I'll post PBK-1 and REW graphs when I start doing the room measurements...


I've played around with the Class A/B vs. A settings on the Emotiva XPA-1L. Realistically I doubt I will need much beyond 30W of power through the efficient S8 speakers so I expect the amps will remain well within the 35W Class A limit (these are 250W monoblocks in A/B). So far, I cannot say I hear much of a difference although I have not specifically done any "serious" listening in Class A mode yet. It certainly does get quite warm (somewhat uncomfortable to touch) after an hour in Class A mode - as expected. Makes for decent space heaters through the holidays I guess :-).

Signature SUB1 clearly visible.
I thought I'd end off with a couple of recommendations for multichannel lovers; both of these titles are only available as DTS-CDs from the early 2000's. I've since ripped these disks and converted to 5.1 16/44 FLAC with the DTS plugin for foobar2000.

- Lyle Lovett - Joshua Judges Ruth(2002 DTS release): Folks, this is a great example of what a good multichannel mix sounds like. Lyle's voice is mostly centered up front, good use of surrounds for ambiance, some discreet vocals in the rear tastefully done. Great dynamic range of DR16. I've always enjoyed the track Church from this album (and used it as a test track for the old MP3 test). In the multichannel version, you literally feel immersed in the choir when they start singing! It's unfortunate that a proper DVD-A/SACD was not released for this album given how good this sounds (already in the DTS-CD incarnation, this album puts to shame many DVD-A and SACD multichannel releases).

- Alan Parsons - On Air (1996 DTS release): Hey, it's Alan Parsons who knows a thing or two about good sounding audio... Progressive rock was made for multichannel - especially so when conceived from the start for surround sound. Rear channels utilized aggressively on some tracks along with birds singing, and cool jet flyby special effects (check out the first track Blue Blue Sky). Again, a multichannel DVD-A/SACD release would have been phenomenal.

Until next time... Enjoy the tunes, wishing you and yours a wonderful Christmas & New Year season.

------------
(BTW: I got my Nikon D800 back from Nikon Canada for repairs on an autofocus issue under warranty. Wow. The focus seems to be spot on and only minimal lens fine-tuning is required now. There was quite a stir online about poor left auto-focus point accuracy as well which seems to be much better now. If you have a D800 and are running into focus issues, check if Nikon can do a tune-up.)

MUSINGS: Those "next generation" game machines - PS4, XBOX One, Wii U...

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Battlefield 4 - PC gaming time in the living room... Silverstone LC14 case in lower shelf. Arcade Street Fighter stick to the left. Old SNES still in the box to the right :-)
As I opened boxes and put things into place after the recent house move, I pondered about the living room situation.

I have a decent sized 46" LCD/LED TV there, my old Denon AVR-3802 receiver, the Squeezebox Touch, and a couple of old Tannoy MX2 bookshelf speakers. It's also where I will have the game machines - the good old XBOX 360 and Wii from a few years back mainly for the kids and the odd Kinect dancing game when friends come over :-).

Looking back, I basically grew up with computing technology... My first computer was the 5KB VIC-20, then Commodore 64, then Commodore Amiga before jumping over to the PC world in the mid-90's putting together my first PC in the venerable AT form factor. All along, games were the programs that truly utilized the computing power of the machines whether it was through hand-entering games published in the old Compute! magazine or being blown away when I first saw the "smooth" character and parallax animations in the Amiga game Shadow Of The Beast! Unless you're using the machine for frequent graphics rendering, or maybe folding, it's probably a safe bet to assume that it's the gaming software and the virtual worlds they create that will reveal the true power of the machine.

For video game machines, I think I've owned at least one representative from each generation. XBOX 360 & Wii, before that XBOX & PS2, before that PS1 & TurboGrafx-16/Duo & Panasonic 3DO FZ-1, before that Super NES, before that Atari 2600, before that Atari Pong (my dad got it as a novelty back in 1977 or so). But looking at the current offerings (a friend already has all 3 of these machines for me to try at his house!) - Playstation 4, XBOX One, and Wii U - I really have no desire to own any of these. I dunno, maybe I'm just getting old and tastes are changing... :-)

I suspect one thing that is changing for me since having kids a few years ago is just the time available for gaming (among other hobbies like audiophilia!). The push I see in this generation of gaming is that of extending the "social" experience. The opportunity to see what friends are doing, which games they're playing, sharing gameplay videos, and of course the ability to play online at the same time. I think that's cool and certainly for those who are looking for that experience, there's probably no better than the unified system that XBOX Live (which I used to subscribe to) and PlayStation Network have available (I've never tried Nintendo Network).

I don't know about you guys, but I'm feeling a bit of "Social Network Fatigue" (SNF) these days though... From the barrage of E-mails, phone texts, to FaceBook notifications, to LinkedIn, to Twitter tweets; I think I'm "good". Friends know how to contact me and I them... I'm not sure I need yet another network to join; especially one which is fee-based subscription and forces people to choose "sides" based on hardware preferences which is ultimately about securing financial revenue (isn't it always? here's a cute South Park take). Certainly you know this was what Microsoft must have been thinking when it first announced that the XBOX One had to be online for gameplay and also threatened to prevent the sale of used games early on. Thankfully, they later retracted this policy.

So, if we take a step back from the whole social gaming scene, what do we have left? The same thing as we've always had... Competing hardware platforms trying to provide the best interactive entertainment content either through inherent hardware superiority or exclusive games. And this is where I'm quite hesitant to buy in at this point.

For those who haven't read up on it, here are the technical stats: IGN Comparison and this from ExtremeTech.

In this generation, AMD's Radeon GCN rules in the graphics department. Every one of these machines is based on this internal graphics architecture which allows a much easier comparison of the graphical prowess. And in the graphics department, without dispute the PS4 is king. If one's priority is the potential to create the most detailed, smoothest gaming experience, then PS4 is the winner - especially that unified 8GB of GDDR5 RAM has thus far not been done and promises some amazing speed and texture quality. Already, with multiplatform 1st generation games like Battlefield 4, this has proven to be the case. Nintendo always seems to march to its own drummer and this is no different with the Wii U; it really cannot compete based on graphics hardware, and as always, must depend on first party titles (talk about proprietary hardware, the disc drive can't even play Blu-Ray movies for crying out loud!).

As for the CPU in these machines, it's very hard to get excited about those 8 AMD Jaguar cores in both the XBOX One and PS4 (the Wii U's PowerPC CPU a.k.a. Espresso is considerably weaker). Speculation is that already the OS is large with a couple of CPU cores unavailable for gaming use in both the PS4 and XBOX One. As to whether this might be a significant limiting factor to the gaming experience, I guess we'll just have to see (the Jaguar CPU core in Kabini is significantly slower than the Trinity core in the A10-5800 APU I've been running for about a year in my HTPC). Obviously offloading tasks to GPGPU streams can alleviate some of the CPU burden but it remains to be seen how this might also hinder the graphics horsepower.

Of course, currently we're only able to review first-generation games and as we have seen in previous console generations, things will only get better in time. With a specific target hardware, custom OS and APIs that can access abilities "closer to the metal", a lot of power can be squeezed out in optimizations. However, I suspect that since the XBOX One and PS4 are based on hardware already in existence in graphics cards for a couple years (GCN has been out since January 2012), the estimates of computing horsepower should be quite accurate and the familiar architecture will lead to optimizations sooner than something like what we saw for the PS3 and its unique "Cell" processor. Furthermore, I speculate that much of the programming optimizations probably will benefit across platforms due to these inherent hardware similarities, multi-core CPU optimizations should become standard for example.

As you can probably tell based on the tone of this post, I think I'm just going to wait on getting a game machine... Looking around here, I already have a number of pieces I can use... An old Silverstone Lascala LC14 case, 500W power supply, Blu-Ray SATA reader, XBOX 360 wireless controller receiver, nVidia GTX 570 from 2011 (still able to play a mean game of Battlefield 4 at 1080P)... On sale I got an AMD FX-8320 + 8G RAM + motherboard + HDD for <$350. Enough to put together a decent gaming rig for the living room which doubles as a general machine for fast NetFlix, streaming video playback off the NAS, etc. I've already got a few older games like StarCraft II, Battlefield 3 and Street Fighter IV which could be fun in the living room and maybe try out a few game downloads off Steam over the holidays. Heck, I might even go online to have a look (I think Battlefield 4 can handle up to 64-players). Another nice thing about PC gaming is compatibility with older titles numbering in the thousands; not to mention using emulators to run those old nostalgic ROMs I grew up to love (hey, my son loves Pacman). One thing I wish could be done easily would be to use the PS4's DualShock4 in a PC game machine; I love the feel of that controller. The Steam Controller looks interesting (probably released in 2014) - for now, I'm quite happy using the XBOX 360 remote controllers.

No matter what, merry Christmas everyone! I wish you all the warmth of friendships and family. To you gamers, I hope you find something you love under the tree on Christmas morn especially if you've been good boys and girls over the last 7+ years waiting for the next generation of consoles...

Till next time... Enjoy the tunes and games :-).

[Update Dec. 29, 2013]: I just watched the documentary Indie Game: The Movie. Wonderful snapshot into the world of independent gaming! I really enjoyed the quirky Super Meat Boy a couple years ago :-).

MEASUREMENTS: USB Cable Extension with Ethernet Cables - does it worsen jitter?

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I mentioned in my "look at the sound room" post the other day that in the Emotiva XSP-1's multichannel "HT Bypass" mode, I'm dealing with some noise when the USB DAC (TEAC UD-501) is connected. In an attempt to fix the problem, I've tried a number of ways to minimize the problem - different USB ports on the motherboard, USB3 vs. USB2 ports, different USB cables, ferrite cores attached to the USB cables, even put in a USB3 PCI-E card... I've even dissected an old USB cable to disconnect the 5V line but it looks like the TEAC needs this present. All to no avail - the high pitched whine and buzz continued.

I then began looking at alternate options; here are a couple I found. The Firestone GreenKey looks interesting. I'm just not sure if it supports high-speed USB2 however which is important since the TEAC needs high speed for the 24/192+ PCM and DSD/DoP sample rates. There are other galvanic isolators like this one but they're just "full speed". Another option is with a better USB card like this Sonore SOtM PCI/PCI-E to USB which is often talked about on sites like Computer Audiophile. However, I really don't see myself spending $400 (with taxes and shipping) for a single port USB card; and I'm not even sure this will do the job I need! (Anyone with one of these cards can comment? A review like this one is useless for my purpose when there's really no discussion of whether they tried it in a noisy system and if it reduced audible background distortion.)

Hypothetically, I thought it might be interesting to try a different angle with this... Knowing that the ethernet system uses isolation transformers, what about the USB extenders using ethernet cables? Already, a cheap one was taken apart on Project Gus and it looks like these devices do indeed use standard LAN filters like the LFE8423. [My mistake, that link was for a USB-ethernet adaptor, not a USB extender like this one.]

With a quick search of eBay (here's a current auction), I decided to take a chance on one of these:
It includes a transmitter (computer end) and receiver (TEAC DAC end). The transmitter side will use the computer's 5V rail and a standard 5V power supply in the plastic bag is used on the DAC side - this of course isolates the power from the computer to the DAC. It also comes with a short USB A-A male cable for the USB connection from the computer to the "transmission" unit.
Notice the 5V power connector on the unit to the left for the receiving device end.
Total cost was ~$60 for the set including shipping from Asia. The unit is capable of 100m (~300ft) transmission distance and specifically rated up to the 480Mbps high speed spec. Unfortunately I don't see an easy way to crack open the little boxes to see what's inside and I figure I didn't want to risk breaking them at this point. Set up was easy and intuitive since there's barely any English in the instruction pamphlet. Plug 'n' play, no drivers to mess around with at all.

So... Did they work for my purpose? Yeah... To some extent.

They did definitely filter out the "computer noise". I can no longer hear the beeps and buzzes when the computer accesses the hard drive or when the CPU is busy processing. Unfortunately the high-pitched whine is still audible when I stand against the speakers; but much reduced - maybe 25% of previous and almost inaudible at my listening position about 9-10' away (ambient background noise is very low in my basement sound room). I tried different Cat5e and USB cables but this didn't make much different. The next time I go to the computer store, I'll see about getting Cat6a STP (Shielded Twisted Pair) cables and maybe shorter 3ft USBs to minimize any potential to pick up interference... I doubt this will make a difference however because I suspect the noise is embedded in the signal from the computer itself rather than picked up due to lack of shielding.

However, the opportunity presented in buying this item is that I can measure what happens to the DAC audio output through this USB signal conversion. If one believes that a digital cable makes a huge difference, then doing this should really create some nasty effects compared to having just a straight computer --> USB DAC cable. We could see diminution of dynamic range if doing this adds more noise to the DAC and furthermore, let's see if the dreaded jitter gets even worse! (Surely it must be awful, right?)

Setup

As usual, here's the setup used to measure the TEAC DAC output when I'm using the USB extender:

Win8 AMD A10 HTPC --> stock 3' USB (supplied in box) --> USB Extender Box --> 50 feet Cat5e cable --> USB Extender Box --> 6' Belkin Gold USB cable --> TEAC UD-501 DAC --> generic shielded RCA --> Emu 0404USB --> shielded generic USB --> measurement Win8 computer

For the direct USB situation, I used a 12' generic shielded USB cable (similar in build to the Belkin Gold USB cable) direct from the Win8 HTPC --> TEAC UD-501.

Notice how rather "unfair" this setup is for my needs... In real life I only use a 6' run of Cat5e cable, but to exacerbate any issues, I'm using a 50' run of just generic ethernet cable.

 

Results

Here's the RightMark summary results. I tested 16/44, 24/96, and 24/192 for your consideration:

Notice how there's no difference whether it's a direct USB cable or the much more complicated USB-Cat5e extender setup. 16/44 as usual measures perfectly and there's not even a hint of loss in dynamic range, worsened noise, or excess distortion using the Cat5e hardware!

Frequency response at 24/192 (16/44 and 24/96 look to be exactly the same as well [not shown]):

Noise level at 16/44 and 24/96 (24/192 looks no different as well [not shown]):

No difference!

How about the dreaded jitter? As usual, let's fire up the spectrum analyzer and have a look at the Dunn J-Test output off the TEAC DAC.



Yet again - essentially no difference.

Summary

Okay, short and sweet - using a USB cable extender with a Cat5e system like this one does not degrade the audio output from the asynchronous TEAC UD-501 DAC. Even measured at a length of 50 feet (much more than my needs), there's no problem.

Subjectively I hear no degradation in the sonic output either and as noted above, it does reduce the noise with the analogue multichannel home theatre bypass on the Emotiva XSP-1 so it does seem to filter out some noise originating from the USB port... Not perfectly silent yet in my system but I'm slowly getting there :-). (This noise does not affect my stereo playback at all off the Squeezebox Transporter or TEAC UD-501, just when I listen to multichannel.)

I had a listen last night with some hi-res stereo 24/192 material - John Coltrane's Blue Train Classic Records 2001 HDAD, Neil Young Harvest off the 2002 DVD-A. Also had a listen to some DSD128 - samples from 2L's website. As usual they sounded very good off the TEAC. All the precision I had come to expect from the TEAC with standard USB cabling was there with no hint of resolution loss or evidence of digital error.

Basically, I'm stumped when I read comments on places like Audio Asylum by audiophile "audio engineer" folks who claim that even adding a ferrite core to a USB cable will do terrible things like worsen jitter (supposedly audible!). As far as I can tell, these folks also never seem to throw up a few measurements or describe what method they use to come to such a conclusion... Even if doing this worsened the sound quality, why do they blame jitter as the problem?

As far as I can tell, bits are bits; at least with a decent asynchronous DAC (I suppose there could be problems with old adaptive isochronous units). Asynchronous USB protocols like the one used with the TEAC work as they should to clean up any stray timing errors so long as the buffer isn't over run and the data is "bit perfect" (easily verifiable with a USB hard drive attached and looking for data corruption). Hard to imagine than anyone needs more than a decent (inexpensive) USB cable for quality sound if doing this kind of cable extension doesn't lead to sonic degradation. I do believe that there are many mysteries in this universe outside of the science we understand today, but digital reproduction in the audible spectrum doesn't need to be one of these! As usual, feel free to provide a link to some results if there's evidence to suggest I'm incorrect.

As we wind down the last days of 2013, I want to wish you all a Happy New Year. And of course a healthy and prosperous 2014 ahead...

Enjoy the tunes.

[2013/12/31 Update]

So, I headed off to the local computer store earlier this morning and bought some 50ft Cat6a STP (shielded) ethernet cables to try. I figure if this doesn't make a difference, the higher quality cable can be used for longer runs around the house.

Surprise! It worked... The whine is now gone. The analogue "HT Bypass" is still a bit noisier than straight balanced cable into the Emotiva XSP-1 preamp but at least there's absolutely no high-pitched noise audible with ears right up to the speakers now.

Again, subjectively, everything sounds great comparing straight USB to the USB-Ethernet extender through the TEAC DAC. Objectively, no evidence of deterioration and looks the same as previous results above:


At least for me, it looks like the use of one of these high-speed ethernet cable extenders is a good option to try for those suffering from USB-related noise in the audio system with no evidence of signal degradation (including jitter with an asynchronous DAC).

Happy New Year and stay safe with those late night parties! :-)

Room Measurements First Steps (Stereo): Paradigm's PBK-1 and Room EQ Wizard...

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PBK-1 mic about to record some subwoofer beeps and boops...

Alas, the Christmas & New Years holiday season is behind us. Each year, I'm amazed at just how quick everything goes by... Never enough time to enjoy the music among the rest of life's other demands.

I didn't get much time off but I did manage to try out some room measurements in the evenings when the kids were in bed.

For all the time I spend thinking about audio components, it's unfortunate that I've spent too little on the room itself. With the new house and room, I hope to change this. Remember folks, as much as "we" (audiophiles) might obsess over the quality of our DACs or (some) might sweat over things like (supposed) cable differences, without question, the quality of the transducer/room interactions trump essentially any of the other factors in the audio system given the quality of electronics these days. Frequency variations in the audio room can fluctuate wildly, on the order of magnitudes of difference compared to the rather trivial differences between DACs for example. It might not be cool or sexy to speak about bass traps, diffusers and absorbers versus DSD or fancy cables made of Unobtanium, but this is what's ultimately going to make huge differences.

Of course, room measurements are not a new topic... Many excellent web pages have been devoted to this. For room treatments, check out Ethan Winer's RealTraps site. Also, Mitchco's excellent room calibration article not long ago documented his procedure nicely.

I'll have the years ahead to add hardware around the room, but as a first step, I'm just going to try the simplest of "room corrections"; EQ'ing frequency response using DSPs.

I. Paradigm PBK-1 (Perfect Bass Kit)

Paradigm, Anthem, and Martin-Logan are owned by the same parent company and share similar technologies in terms of room correction. The Paradigm Signature and Martin-Logan subwoofers have built-in DSP units that can be programmed using a PC with the PBK kits; based off of the Anthem Room Correction algorithms. As the name implies, this kit is used for bass correction only as it applies to the subwoofer. Audio levels still should be checked with the rest of the system in order to make sure the sub integrates well.

As of this writing, the current version of the PBK software (2.01) is not compatible with Windows 8 (even with compatibility settings)! I had to dual boot the old laptop used for measurements with Windows 7 and run the software through that... Paradigm, please update the program!

The software was easy to run and essentially self-explanatory. A USB cable connects the included PBK microphone (shown above) to the PC, then another USB connects the PC to the subwoofer. Each microphone has been calibrated and identified by serial number in the software. The program is capable of measuring multiple locations (max. 10) in the room to smooth out the bass response. Since I'm most concerned about the "sweet spot" and want to limit the potential of suboptimal calibration, I took 5 readings all around the central seat and the 2 adjacent seats. The program then will run thorough the calibration algorithm and show a screen that looks something like this...


You have the option to adjust the DSP crossover frequency point which in the graph above I've set to 160Hz (default is 250Hz) as well as how steep the filter should be. This gives you some customization options (not much).

The red curve was what I got in the room. As you can see, I have quite a dip at around 65Hz (red) which was correctable to some extent (purple). The peaks (eg. around 30Hz correlated to the calculated lateral mode for the room size at 29.1Hz using this online calculator) were easier to correct, and it's nice to see good frequency response down to 20Hz with this sub. (Check this link out for the room mode math calculations.)

The program will automatically upload the new settings to the subwoofer and away you go... Very simple calibration to do.

II. Room EQ Wizard (a.k.a. REW)

REW (5.01beta) can be downloaded free off the Home Theater Shack website - just need to register. It's just an amazing piece of Java code for the audio enthusiast.

About 3 years ago, I purchased a calibrated Behringer ECM8000 mic (I see they don't sell these any more at that site). This microphone has served me well over the years and put to good use here again (note that I actually measured it a little lower at ear level sitting on the couch than the picture below):


Using the EMU 0404USB as measuring ADC, here's the raw room response with just the PBK settings in place for the subwoofer integrated with the main front speakers (1/6 octave smoothing) using the TEAC UD-501 DAC:

Although I don't know if I fully trust the Behringer mic below 30Hz and above 15kHz, it's good to see frequency response down to 15Hz. Again we see the room mode around 29Hz. The deep blue line represents the eventual target curve we're aiming for based on the default REW house curve (for those looking for the excitement of a bass-induced thrill ride, try this target curve). For those looking for more based on home theater wisdom, check out this link on House Curves and more!

Letting REW perform its own EQ from 20-200Hz plus a few small adjustments on my end resulted in this mathematical prediction of room response:
Much more controlled on the low end using 7 parametric EQ settings (you can see the numbers above) plus 2 settings at 3kHz and ~11kHz to roll off the top. Total of 9 EQ settings were programmed into the Behringer DEQ2496 applying the adjustments in the digital domain and looped back to the Transporter for DAC duties.

Since some frequency boosting is involved, I reduced the DEQ2496 digital output levels by 6dB and double checked with some really LOUD music to make sure the EQ settings did not lead to clipping. The 1997 Iggy Pop insane remaster of The Stooges' track "Your Pretty Face Is Going To Hell" off Raw Power is a good one - average dynamic range value of 1dB for the album! If the DSP processing doesn't clip with that track, it's probably not an issue with >99% of my music.

Since it's always good to confirm that the EQs are actually doing what they're supposed to, here's the actual measured room response with & without the Behringer DEQ2496 played through the Transporter as DAC (measured at a higher level on a separate day):
Before REW EQ:

After REW EQ:

OK. Nice real-world confirmation that the EQs are doing what they're supposed to. The predicted results checks out even with a different DAC (measured with TEAC, confirmed with Behringer & Transporter - I knew from previous measurements that the Transporter is very close in frequency response to the TEAC)!

Looking Ahead...

Like I said above, I have many days ahead to make this room "work" better acoustically. EQ'ing so far is just the "quick and dirty" first step at this point focused essentially just on volume equalization (although I guess the PBK might be doing more in the algorithm). I haven't even begun to try using the Audyssey MultEQ XT for multichannel yet (in the Onkyo receiver), nor room treatments...

Regarding room treatments, there's much to do! Here's the measured waterfall spectral decay plot in REW:
15Hz - 20kHz
20-200Hz
With EQs in place, the decay time of course remains high:
15Hz - 20kHz
20-200Hz

Plotted at 500ms duration down to 40dB with 1/6 octave smoothing, I'd really love to see more uniform steeper decay (<<300ms). Bass traps in the corner and absorption panels to the sides at the first reflection points could do the trick. Hmmm, maybe this will be a project for Spring Break - whip out the saw, stapler gun, make some wood frames, grab fabric and a stash of Roxul Safe'n'Sound :-).

There is also the issue with time alignment as addressed by mitchco using Acourate and convolution filters. That's another level of tuning I'll have to leave for another day! So much to do, so little time...

For now, the subjective sound quality has improved. There's already notably better control to the bass notes from Rebecca Pidgeon's "Spanish Harlem" (to use a well known audiophile favourite). Time to just sit back, relax, and enjoy some tunes! I've got a couple of ideas for tests coming up.

Musical selections recently:
- Daft Punk's TRON: Legacy soundtrack (2010) on Blu-Ray was stunning! I missed the movie in the theaters and rented the 3D Blu-Ray the other day... The movie itself was OK but the surround effects and techo score really made the movie an audio feast.

- The Eagles. Love 'em or hate 'em, I reacquainted myself with the multichannel DTS version of Hell Freezes Over (1997) the other night. IMO another fantastic multichannel release from the earlier days of surround sound when DTS was releasing their DTS-CD's (this was also my first concert DVD). That live ambiance really shines through as if you're sitting in the audience that night. The guitar work and percussion sound great in the new system on "Hotel California" - especially the bass impact. I saw them in concert about 3 years back and that too was a blast.

- Speaking of bass... I don't often buy modern pop recordings but I did enjoy listening to the recent album by Lorde - Pure Heroine. Fantastic job by the 16-year-old from New Zealand. The current Top-40 'hit' "Royals" gives a nice taste of the cavernous bass found throughout the album ("400 Lux" is another to check out). Have a listen to this album through a system with clean bass down to 20Hz and see whether you think you need a subwoofer :-).

Addendum:
In the "Answer To What Question?" and "What Were They Thinking Of?" files... I was looking at the recent CES2014 announcements in the usual audiophile watering holes and found this:
Esoteric Grandioso D1 Monoblock DAC
Given the level of performance of even modest DACs these days, I really can't imagine what would be the reason to go monoblock with a DAC. Seems like doing this could make things worse (channel desynchronization? need for DAC matching?) and at a significant expense ($22,000 each, not to mention all those extra cables!). Small price to pay to feel grandioso I suppose. As usual, would love to see the measurements for a pair...

MEASUREMENTS: Google Nexus 5 and Nexus 7 (2013) audio quality...

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Mobile space is where it's at these days for major consumer computing innovations... New advancements in wireless communication, low power, more speed. Amazing just how fast progress is being made! It was only in early 2000 that the 1GHz barrier was broken for consumer desktop CPU's, 2005 was the first mainstream dual-core x86 CPU (both AMD X2 and Pentium Extreme 840). Now, we've got >1GHz quad-cores with 1080P screens on handheld devices in less than 10 years... (Yes, I know an ARM CPU isn't directly comparable to the x86, but still impressive nonetheless!)

For fun, I wanted to run the current Google Nexus' headphone output through the measurement gear to have a look at what some mobile folks are listening to...

I. The 'Victims'

A. Google ASUS Nexus 7 (2013 2nd generation model) - Android 4.4.2 "Kitkat", stock ROM, 32GB storage

 

I'm really liking the smaller 7" form factor for tablets. This tablet acts as a fantastic controller for my Squeezebox Server using Squeeze Commander (my stereo PCM library streamed to Squeezeboxes), Gizmo for JRiver (all my DSD music to TEAC UD-501), and FoobarCon Pro for foobar2000 (multichannel PCM to ONKYO HDMI receiver).

It feels fast with a 1.51 GHz Qualcomm Snapdragon S4 Pro. Have had it for a few months, but I've actually never listened to the headphone output until tonight. Sounds fine with my Audio-Technica ATH-M50 in terms of bass weight and resolution. It can't drive the M50 loud so you'd want to use higher sensitivity headphones.

B. Google / LG Nexus 5 - Android 4.4.2 "Kitkat", stock ROM, 32GB storage

Nexus 5 lying on the Panasonic QE-TM101. Love the QI wireless charging!

The nice UPS man dropped this off on Christmas Eve. I gotta say, compared to most of what we see in audiophile land, this is true value! Plenty of features, one of the fastest current mobile CPU's, great 1080P screen as well squeezed into 5". This phone has replaced my Samsung Galaxy S2. Some people have complained of poor battery life; I find that it's fine with Google Now turned off.

Like the Nexus 7 above, I didn't have a listen with headphones until tonight. Maximum volume seemed a bit higher than the Nexus 7 using the ATH-M50 which was a surprise. Again, in sounds pretty good. "Dead Already" off the American Beauty soundtrack maintained its usual bass impact through the headphones. Demanding loud tracks like "To Victory" off the 300 soundtrack wasn't as defined but not bad.

II. Results

Okay, pretty straight forward setup here... Android accepts FLAC without issue, all that was needed was to copy over the calibration and test files via USB and off to the test bench. Of course, all DSP/EQ/bass boost off. I left the phone and HSPA+ data on for the Nexus 5 and WiFi on for Nexus 7 as the most likely situations in daily use.

Nexus device headphone out --> phono-to-RCA cable --> EMU 0404USB --> shielded USB --> Win8 laptop

Unfortunately I don't see any specs on the output impedance for either of these units so it's hard to discuss headphone matching.

RightMark summary:

Totally unfair, but I threw in a couple of 24/96 Squeezebox measurements on there - the SB Touch which represents a good upper level consumer device, and the Transporter which is in the audiophile league of audio performance.

16/44:


24/96:

Jitter:


III. Summary

Well, there's not much surprise here. The built-in DAC's on these devices are limited in fidelity. The DAC circuitry is integrated into the tightly packed SoC which includes the CPU and GPU, situated in close proximity to wireless communication hardware for WiFi, BlueTooth, G3/HSDPA/LTE transmission...

Having said this, I was actually a little surprised to find that the smaller Nexus 5 phone performed a little better than the Nexus 7! The frequency response was more even and there was less harmonic distortion found... Evidently, it's not the size that matters.

Neither unit could benefit from 96kHz sampling rate - looks like it's downsampled to 44kHz.

Although neither device could deliver beyond about 16-bit dynamic range, it was interesting to find that 24-bit data resulted in slightly lower noise floor (you can see this easily with the J-Test graphs). Interestingly, the jitter modulation pattern was visible with both devices for the 24-bit J-Test suggestion high jitter levels (but really, who cares?).

Obviously, both the Squeezebox devices measure and would sound superior to these portable units.

I've found it curious the recent developments in smaller audiophile gear. Those small USB DACs for example seem to preform reasonably well - the AudioQuest Firefly, Meridian Explorer both look good and measure well for those with small desktop space or want a high quality DAC on trips. It's quite clear the success of the Light Harmonic Geek speaks to the market in such devices.

What I'm not so "bullish" about are those "audiophile iPods"... Stuff like the HiFiMan units, or the Iriver Astell&Kern. I find it hard to consider portable players as anything other than convenience items. They're meant for headphone use of course, but even if they performed excellently (it looks like the Astell&Kern AK100 measures well), what's the point? The more resolving headphones that can benefit like the Sennheiser HD800 are open design with poor noise isolation and I cannot imagine many folks crazy enough to walk the streets or take the subway with them on. To make matters worse, the high end headphones usually require more power and that's going to really drain battery life assuming the "audiophile iPod" even has the oomph to provide enough volume in the first place.

Bottom line... For the purpose of the commute, these measurements of the Nexus 5 and 7 appear totally fine assuming you have a good pair of efficient headphones. From my collection, volume was adequate with the Apple white earbuds, JVC HA-FX40 IEM, Sony MDR-XB500, Koss PortaPro, but not really good enough for the Sony MDR-V6 or aforementioned ATH-M50 in a busy environment... Don't even bother trying open headphones like the Sennheiser HD800 or AKG Q701. However, surprisingly the Nexus 5 drove the Audio-Technica ATH-AD700 better than I thought.

DEMO & MEASUREMENTS: What does a bad USB (or other digital audio) cable sound like?

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Okay, so the other day I was installing my new BenQ BL2710PT monitor (reasonable monitor for a decent price) and as I was rummaging through my old cables, came across a very oldUSB 6' cable that I probably got free when I purchased an old Samsung laser printer back in 2001 during the transition from USB 1.1 to early USB 2.0.

This cable is the thinnest, most flexible, likely most poorly shielded USB cable I have; in other words, about as "bad" as it gets when connected to a quality USB DAC which expects to operate in high speed USB 2.0 mode without completely failing... Behold the "Bad Cable":


Plugging this cable into my desktop ASUS Essence One provided the opportunity to demonstrate just what a poor USB cable does to the sound... I'm sure this is "old hat" to those who have experience with digital audio, but for those who haven't, have a listen...

I recorded 1 minute of a freely available track from Jason Shaw called "Pioneers" from here off the Essence One fed into my EMU 0404USB to the usual Win8 laptop using a good quality cable versus the flimsy one above.

Good USB 2.0 cable - well shielded 12', ferrite core on both ends of this specific cable:


"Bad" USB cable as pictured above - poorly shielded against interference and incapable of transmitting at bit-perfect high-speed data rate to the ASUS Essence One:


Even though SoundCloud recompresses the uploaded FLAC audio, I'm sure you can appreciate the obvious errors in the "Bad Cable" sample. (You can press play back & forth between the two samples to A-B them if you want.)

What you're hearing is what happens with digital error (ie. not bit-perfect), similar to watching digital TV with the occasional data error leading to macroblocking and bad pixellation as in this sample found off Google (notice the blue stripe due to digital error):

It's worth noting a few characteristics of this poor cable as it pertains to sound:

1. Poor digital cables leading to digital errors sound like brief pops or occasional static (assuming they do not completely malfunction). They're similar to the errors you get when ripping a CD without something like EAC or equivalent. Sometimes, you'll hear very brief dropouts. Depending on the data packet disrupted, occasionally they will occur in only one channel but not possible for this to happen consistently in a single channel. Remember that although asynchronous DACs have the capability to buffer, hence improve timing and lower jitter, they do not (at least not in the case of the Essence One with the CM6631 USB interface as far as I can tell) necessarily error correct or ask for a packet resend. The more data error, the less the amount of "normal sounding" music will be heard. Obviously if the data error occurs every few minutes, it might be difficult to detect, but if it happens frequently, it's not subtle.

2. A poor digital cable does not result in overall level changes in the song... This is not like analogue distortion that can consistently alter the volume level or change the dynamic range uniformly or periodically.

3. Similar to the above point, poor digital cables are not capable of changing the overall tone of the sound. There is no such thing as a digital cable capable of acting as a "tone control", making certain sounds "brighter" or "warmer". A passive digital cable is not capable of acting with some kind of frequency filtering mechanism.

4. Poor digital cables do not consistently do anything to the soundstage. A poor digital cable cannot make a voice or instrument sound "distant" or move it "forward", or pan the soundstage to the left or right as a whole or in relation to other components of the music.

5. Bad cables cannot cause speeding up or slowing down of the data transfer. Poor digital cables therefore cannot cause sporadic or consistent timing issues like warble (speed up/slow down pitch changes), "pacing", or rhythm problems.

6. The concept of cable "break in" makes no sense with digital audio cables. If it carries data accurately when plugged in then the only problem that can happen in time is corrosion at contact points or reactions such as oxidation of the metal over time. This can only lead to transmission errors as demonstrated above, not some magical improvement due to "break-in".

7. I was reminded here the other day about the measurements with a poor RCA cable I used as coaxial SPDIF last year. Indeed, if you use a very poor, unshielded RCA cable paying no attention to the expected 75-ohm impedance specification with an SPDIF digital interface that's not galvanically isolated (eg. coaxial SPDIF of the ASUS Essence One in that case), noise can be introduced into the system. However it does not take extravagantly priced cables to make things right (an inexpensive 6' <$20 decent shielded cable from a reputable company will do). As always, noise can be introduced into the analogue domain with any electrical connection (or just being careless like putting your DAC right on top of a noisy desktop computer), so it's not really an issue with the digital system itself.

You might be curious how the 2 USB cables measure in terms of jitter...


Surprised? As you can see - not much difference at all! If you monitor the realtime FFT for the J-Test, you will see errors "popping up" with the bad cable due to bad data transfer, but in between, the jitter plots are essentially indistinguishable! This is expected... For an asynchronous DAC like the ASUS Essence One, jitter rejection is handled very well by design and there's nothing the passive cable can do about that.

Now I'm sure there will be a number of folks who disagree and hear various effects in the list above (see here, here and here for some interesting perceptual accounts and/or creative writing). The thing is, where is there decent evidence to show that passive digital cables (and I'm talking here not just of USB but also the SPDIF variants like coaxial or TosLink) sound different if they're error free (a.k.a. bit-perfect) and built to specifications (assuming no issue with analogue noise as in item 7 above)? I've never seen manufacturers come up with anything of substance... Or hobbyists/DIY guys show/demonstrate verifiable claims... As usual, I'm happy to change my mind if some kind of objective evidence exists since I personally have not subjectively heard a problem except as demonstrated above with digital errors.

(Digital cable summary from a number of months back for those who might have missed it. Recent post on EETimes blog on this.)

Recommended album:
- Have a listen to Babatunde Olatunji's posthumous 2005 album Circle Of Drums. This is one of the best Chesky albums I have heard. The drums sound fantastic with wonderful tonality and sense of 'space' around the instruments; a lovely exploration of African drums and rhythm. Unless you believe you can hear the difference between 16-bit noise floor and that provided by SACD, IMO there's no need to buy the SACD because it appears to be a 44kHz PCM upsample (here's the Master List). There is a multichannel mix on the SACD which sounds OK but derived from post-processing. An impressive sounding and quite enjoyable record for those interested in world music nonetheless!

Relax and enjoy the music!

MUSINGS: The Audio PC / Music Server / HTPC (Basics, My PC, and Some Generalization)

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I'm in the process of finishing up some measurements - I think many will find it interesting in the days ahead. I'm going to let that article percolate a little first however. This week I thought I'd spend some time discussing/considering the computer system for media consumption; a bit on both the hardware and software aspects, and hopefully putting together bits and pieces I've done over the past year in the process. As you can imagine, there's quite a lot to cover...

In the hopes of a reasonable summary, here are the 3 main component functions of computing devices (almost everything is a 'computer' these days) in the media room:

1. Media Client or "Media Player" or "Transport (to a DAC)": The computer acts as a "front end" to feed your DAC/receiver/amp and or TV/monitor. Of course you can also have a good internal sound card (like the ASUS Essence STX), and I presume few are using the motherboard's sound output (usually of comparatively poor quality). You interact with the computer with whatever player software to control which tracks/files are played, how to "fast forward", "stop", "pause", etc. in much the same way as the disk spinners. Specialized "computers" may have hardware controls for this like a play button on the front panel of the chassis, maybe a specific remote control unit rather than using a generic keyboard. On-screen display on the monitor/TV can be as simple as a web-based interface like the Squeezebox server or the customary GUI of something like iTunes, foobar, JRiver, etc.

This transport/playback task can also be delegated to streaming devices of course (which are essentially little computers inside). For example, the Squeezebox familyMeridian Sooloos, Cambridge Audio streamers, Naim NAC-N, Linn DS, Bryston BDP-2, LUMIN etc... That's the higher end, but on the low end, you have things like the WDTV Live previously measured. These devices usually need to be connected to some kind of server for one's music (although some devices like the Squeezebox Touch can act as its own miniserver) and many can access Internet based streaming media like the hundreds of Internet radio stations around the world, Spotify, MOG, Pandora, SiriusXM, maybe Beats Music in the days ahead.

Although audio streaming may be all one needs, for those who "want it all", the pinnacle of the media room computer is the HTPC (Home Theater PC) with both audio and video playback capability. The HDMI interface has become the de facto digital audio-video cable to do it all. Multichannel 7.1 hi-resolution audio is essentially universal these days with modern HDMI interfaces, and 24/192 sample rates can be sent to a decent modern AV receiver with no problem. Some AV receivers will also accept DSD. IMO, multichannel PCM is preferable because DSP manipulation of the audio stream is an essential part of getting multichannel right... Good bass management (some SACD players are able to do this in DSD), channel reassignments (eg. 5.1 fold down to 4.1 system), room corrections, are all easily done in the PCM domain and would require a DSD-to-PCM conversion step if you're sending out a DSD bitstream. This limitation of DSD is a big one in multichannel and unlikely to be solved any time soon... If ever...

As for video with the HTPC, it's trivial to achieve 1080P. 4K resolution can be reached with the current HDMI 1.4 specification. As of early 2014, I suspect the market feels little compulsion to buy the current generation of 4K TV's and one would achieve little benefit apart from early adopter bragging rights (and spending quite some money in the process!). First and most painfully obvious, there's no content nor even a clearly announced means of media distribution (it looks like 4K/UHD Blu-Ray is still in development). Second, I'd suggest waiting for the wide availability of HDMI 2.0 which would allow 60Hz 4K frame rates (I see that Sony has released firmware for certain TV models already, DisplayPort can already achieve 4K/60) since there does at least seem to be some push towards >24fps movies and if I'm getting a "next-gen" TV, I'd want that. At this time, the real benefit I see from 4K is finally being able to watch 3D in full 1080P on a passive display. I'm waiting for one of those to hit an affordable price range in the 80+" size & HDMI 2.0 :-).

2. Media Server: This is the "back end" where you store your music (videos, movies, pictures). In this day and age with easy connectivity, there's nothing to keep the server in the media/listening room. Many people have opted for NAS storage and since there's a CPU inside the NAS unit, it could also run server software like Logitech Media Server, or the scads of UPnP/DLNA servers. In fact, in a home where there's wired ethernet throughout the house, you can easily have the server computer or NAS on a different floor/room. In my experience, a gigabit network can transfer data just as fast as many inexpensive high-capacity hard drives (50-100MB/s is normal with gigabit ethernet using standard Cat5e cable). A great benefit to this is that you can keep the playback machine (computer or streamer) simple, low power, cool and silent without having a bunch of hard drives running in the same enclosure while listening to your music or watching movies.

3. Mobile Control Apps: Although not specifically the computer itself, the ability to use one's smartphone or tablet computer has been a great boon to the usability of digital media playback. No longer do we need to turn on the TV to select albums, or select video. These days, I still use iPeng and SqueezePad on my iPad to control my Squeezeboxes (Logitech Media Server). Squeeze Commander works fabulously on the Android devices. Cover artwork adds to that overall presentation.

However you want to mix-and-match the functions above, there are a myriad of options based on what OS you choose, which server software, and how the media is being played. The hardware itself can be any combination of devices like NAS, laptop, desktop, network streamers, etc. It is this fantastic flexibility that can be a source of frustration to those starting to enter the computer audio world. Commercial companies are obviously interested in capturing part of the market with devices such as the Aurender computer systems, Sooloos Music Server System (see Streaming products). Not surprisingly, these turn-key products are usually Linux based, low power, relatively slow (often Intel Atom CPU, sometimes ARM based), and generally quite a bit more expensive than something one can put together with standard commodity parts. The greatest thing about true technological innovation in the marketplace is the deflationary price pressure - take advantage of it if you can! Have a look at Computer Audiophile for some ideas on building one yourself.

As a "case study", I figure it might be of interest to show the system I'm currently running...

I basically have an all-in-one box that's a server to my Squeezebox devices all over the house, a digital audio transport to my TEAC UD-501 DAC, as well as full HTPC functionality to the ONKYO receiver and 55" LG TV for movies and videos.
Hmmm... Maybe should clean up a few of those cables in there. Logitech Unifying receiver sticking up front for the keyboard.
HTPC quietly doing it's thing in the corner... The smaller box beside the computer is a CyberPower CP1500PFC UPS. Keyboard is the Logitech TK820 with touchpad.

Main hardware components:
- Fractal Define R3 midtower quiet case (2 quiet case fans - rear exhaust and front to cool HDs)
- Seasonic X-400FLII fanless 400W PSU
- ASUS F2A85-V Pro motherboard (HDMI 1.4)
- AMD A10-5800K APU (integrated AMD Radeon HD7660D GPU)
- CoolerMaster Hyper 212 Plus CPU cooler
- 16GB (2 x 8GB) Kingston Hyper X Blu DDR3 RAM
- SiliconImage Sil3132 port-multiplier eSATA card
- 128GB OCZ Vertex 3 SSD boot drive
- 2 x 3TB WD Red for music library
- 1 x 2TB WD Red for video/movie library
- 1 x 1TB WD Green for web server data
- 1 x 1TB WD Green for misc data backups
- LG BH12LS35 Blu-Ray reader/writer - playing the occasional Blu-Ray & audio ripping

OS: Windows 8 Pro / 64-bit

The AMD A10 CPU has worked well for me in the past year. In fact, I undervolt and underclock it slightly to 3.5GHz to keep it cool and quiet with the CoolerMaster CPU heatsink/fan. I haven't measured, but the CPU power consumption would be substantially less than 100W.

Notice that the HTPC isn't built with necessarily the newest generation hardware components. In fact, much of this was put together more than a year ago. Unless some "disruptive" killer app were to be released that needs much more computing power, I suspect this would be all I need for the next few years (clearly the push to upgrade is slowing). Audio processing doesn't take much power. At times I will turn on SoX minimal phase upsampling à la Meridian for the Transporter (see VirusKiller's thread), other times play with JRiver's PCM to DSD conversion à la EMM Labs, or try out a convolution room correction filter with foobar - never has the AMD A10 felt underpowered for these tasks. With a better audio room since moving into the new house in late November 2013, I've been quite aware of the slight noise the computer makes. Recently, I replaced the power supply with a fanless model (the Seasonic 400W) which lowered the noise a bit.

At this point, the computer is still slightly audible on account of the spinning hard drives (softer than the Panasonic Blu-Ray player in use). The problem with upgrading the room significantly (my ambient SPL is <30dB(A) at night) is that you also have to get the other parts up to spec as well ;-). I guess I can look at moving the hard drives used for video and web serving over to another computer on the home network to drop a few more dB's.

I have 6TB of space for all the music (all backed up on another machine over the gigabit network). The library consists of stereo PCM CD's (~3000+ albums) and hi-res downloads and rips (~250 albums). I have some multichannel 5.1 music (~100 albums) in PCM format taken off my DVD-A/SACD/DVD/Blu-Ray/DTS-CDs. All the PCM music encoded losslessly as FLAC.

Finally I have a small ~30 album collection of DSD stereo music which I know are either sourced from genuine DSD recordings or analogue transfers (ahem... no Norah Jones Come Away With Me type faux DSD, thanks). These DSD recordings are stored as .dff because I like lossless DST (Direct Stream Transfer) compression. As I noted months ago in my SACD/DSD Musings, I do not like the fact that one currently cannot have both tagging and compression. Without compression, DSD files are unnecessarily large, wasteful, and ultimately inelegant IMO. DST compression brings them down smaller than the size of average 24/96 files and I make sure that the filenames I choose can be easily parsed for album, track number, artist, and title. I don't have many DSD albums on the server so it hasn't been difficult. Despite the ongoing hoopla around DSD, I remain sceptical that DSD will have much traction unless a simple foundational issue like a fully featured, modern, file format is addressed. (Actually, I just suspect there's not going to be maintained traction simply because DSD doesn't bring much to the table...)

I'm not going to say much about the video playback since much of it is just family videos with some Blu-Ray rips I made for demo purposes when people come by to visit. The AMD A10 APU has a built-in graphics processor and the HDMI out of the motherboard works well to send multichannel 24-bit audio and 1080P video to the AV receiver.

Player Software:
I see audiophiles can really get heated about this. For those who have been around this blog for awhile, you'll know that I did not measure any difference between bit-perfect software players whether with Windows or Mac OS X using the TEAC DAC. This was the same for PCM and DSD. Furthermore, the "audiophile" player JPlay made no difference and in my opinion risked audio errors with extreme settings like unnecessarily low buffer space. Since I do not claim to have particularly "golden" ears (I'm in my 40's now), I likewise hear no difference between these players.

On my HTPC, I currently have 3 audio "servers"/players running:

1. Logitech Media Server (aka Squeezebox Server back in the day). All my stereo PCM music is streamed out to the Squeezebox units I have scattered around the house. In the listening room, I have the Transporter on the rack. BTW I do have a few 24/192 albums in my collection, but for most DVD-A rips where I have the physical copy, I usually downsample to 24/96 anyways. There are many "fake" 24/192's out there (just like 44kHz upsampled DSD) and even for those that are genuine hi-res recordings or analogue transfers, there's rarely content related to the music itself above ~40kHz (not that anyone would be able to hear it!), so I figure there's no point wasting space.

Because I have 16GB of RAM, I've set up a 4GB RAM disk (free one for personal use) and modified my configuration for LMS to put the database there. Really speeds up library searches - almost instantaneous.
Logitech Media Server (LMS): Some Billy Joel or Benny Mardones anyone? :-)

As mentioned above, I use SqueezePad, iPeng, and Squeeze Commander to control the server.

2.foobar2000: Fantastic, flexible, free software I keep running 24/7. The library keeps track of all my multichannel PCM music and the default output device is to the ONKYO TX-NR1009 multichannel receiver via HDMI using WASAPI. FoobarCon Pro is my preferred controller software off my Nexus 5 or 7; cool that it also has panels to view artist bios and track lyrics so you can sing along as the surround sound plays :-).

Foobar playing 24/96 5.1 Crowded House DVD-A rip.

3. JRiver Media Center 19: Again, I keep this running 24/7. Works beautifully and fully featured with a huge number of customizations and options. Although it has many advanced features like the ability to realtime upsample PCM to DSD64/128, I'm actually not using this for any of my PCM playback... JRiver is dedicated to feeding the TEAC UD-501 all my DSD music in native ASIO! As I mentioned above, .dff files cannot be tagged natively but JRiver can keep an internal database and has the ability to parse the filenames to "recreate" the album data, sort artists, tracks, etc. Furthermore, it decodes DST without any problems even when in the past I found issue with foobar's DSD decoding plugin. Nice :-). I don't think there's another software package that will do all this in such a hassle-free fashion with a lovely presentation at a very reasonable price (~$50USD).
A peek at the DSD library in JRiver using filenames and path to reconstruct tagging information.

JRiver provides the free Gizmo app on the Android for playback control. It works well but functionality is more basic than something like the Squeezebox controllers above. However, the really cool part is that JRiver can transcode and play music and videos to said Android device via Gizmo. A reminder that JRiver isn't just for music but works well as a full-featured "Media Center" for all your A/V needs.

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So that's a glimpse into how I'm currently using my HTPC. Perhaps some of this could be useful for your setup as well.

Now about audiophilia and computer audio...

To close off this blog entry, let's talk about the computer in an audiophile setup; specifically achieving excellent sound quality. I know many people believe that all kinds of arcane software tweaks such as turning off unused processes like printer services, BIOS tweaks, etc. are necessary to ensure good sound (something like this). Much of the OS tweaks probably do no harm and some of these recommendations may have been useful at one time (like a decade ago); I just don't think much of this is relevant any more or makes any difference. As far as I can tell, jitter under high CPU load is not an issue even with a simple TosLink off a motherboard as I showed here so I hope nobody falls for the "it causes jitter to be worse" explanation unless demonstrable at the level of the DAC output. Others in the past speak of using low power CPU's for audio (I haven't seen as many proponents these days). For me, the good thing about a more powerful machine is a speedier user interface, faster file scanning for the server, and also the opportunity to use DSP like convolution room correction filters without the machine breaking a sweat. Of course, you'd want a more powerful computer for video playback. Some others even advise against using lossless compression. Seriously, does anyone still actually believe a processor unintensive task like FLAC lossless decoding will cause enough electrical noise/interference to make it sound worse than a WAV file especially played off an external DAC!? (I certainly hope ideas like this will become just as bizarre as the belief in greening the edges of CD's 20 years ago.) Sadly, over the years, various audiophile magazines have promulgated much speculation and disinformation without checking facts or consulting with common sense (much less science/engineering).

Let's keep it simple - IMO, the main ingredients of a good computer audio setup:

1. Keep the computer sonically quiet! As few fans as possible if not fanless. Laptops are great for this - something like a MacBook Air or Ultrabook would be fantastic for example given how quiet they run. If you can, relocate noisy hard drive servers to another room with wired network (I consider wireless too unreliable for my taste and can be strained by high-resolution data rates).

2. Keep the computer away from your audio gear to reduce EMI/RF from entering the analogue path.

3. Get a good DAC. External units are great because they can be placed with your other components and isolate the computer as in point 2. Make sure you're using the best driver especially with PCs such as bit-perfect ASIO instead of going through the Windows Mixer to ensure bit-perfect output. Also, jitter has more to do with the quality of DAC than anything you fool around with on the computer side. From my measurements posted around here over the last year (eg. look at the TEAC UD-501 PCM results), a good modern asynchronous USB (or ethernet streaming) is generally better than SPDIF (coaxial/TosLink) due to lower jitter (J-Test results better but for the most part I doubt it's audible).

4. Things to not sweat about:cables - just make sure your power cords and interface cables work and look good enough to you in your room. IMO expensive cables may look good and convey a sense of authority, but please do not equate aesthetic value (eg. jewellery) with function (ie. "better" sound). Specific make/model of computer - again, this is aesthetic and so long as it runs your choice of OS, you're good. (No, I do not consider Apple computers as somehow better sounding.) OS - Mac, Windows, Linux, whatever so long as your server/player software runs well on it. As I showed here, different laptops and OS's connected to the same asynchronous USB DAC results in exactly the same analogue audio output. While I can't vouch for every computer, so long as bit-perfect output to the digital interface is assured, there's no need to fret. Player software - Again, see my bit-perfect measurement posts here, here, and here. Find one that has all the features you need and achieves bit-perfect output.

If you have the above down pat, then by all means tweak to your heart's content! Just don't break anything...

I just realized I've been building my computer audio library since 2004 (10 years already!). For those new to computer audio, I suspect all of this could sound overwhelming (and I'm sure I missed some important points). Stick with it, play around with it - it won't take long to pick up. No matter what I do with the computer setup, without doubt, the most time consuming bit of all has been to make sure all the music is tagged properly and named in a consistent fashion (try Mp3Tag). Keeping the directories clean, using the same filename for cover images, and ensuring bit-perfect rips (try dBPowerAmp CD Ripper) do take time and effort; this is as expected since it's all about the music, right? Despite all the effort, high-resolution digital audio is as good as it gets for the audiophile who values high fidelity and the convenience in accessing all your music with a few search keystrokes is undeniable.

It's a great hobby with many avenues to explore. Just don't forget to listen, and enjoy the music :-).

PS: Backup regularly.

MEASUREMENTS: Power Cable Redux. The Synergistic Research Tesla T2 SE, T3 SE and PowerCell 4.

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As promised over the last months, I've been wanting to borrow some expensive cables to measure to see for myself if I can both objectively and subjectively experience a difference. I've measured power cables before with my ASUS Essence One DAC's output but it just so happened that recently a friend decided to "go deep" into the world of audiophilia and purchased a little "family" of Synergistic power cables to try out:

What we're looking at here is a "tree" of Synergistic gear :-). Plugged into the wall is a "Tesla T3 SE" cable connected to a "PowerCell 4" (basically functions as a 4-outlet power bar). Coming out of the PowerCell on top are 2 "Tesla T2 SE" cables and an Audience power cable (not evaluated here). The blue lights come from the "Enigma Bullets" which I'll address a little later. My friend has been listening with them for >6 months so there's no issue with new cable "break-in".

As has been expressed by others, it's hard to make a case for power cords... The AC in our homes are connected through tens/hundreds of miles of cabling of various gauge. Within one's abode, it's interconnected with multiple outlets (unless of course you hire an electrician for a dedicated line) usually through 14AWG copper wires for most 15A circuit breakers here in Canada. Could the last few feet be significant?! Does running fancy cables like those above really improve sound quality? Here's a chance to have a look and listen...

I. A Look at the Synergistic Tesla T2 SE (5'), T3 SE (5'), and PowerCell 4

For the purpose of these measurements, I wanted to keep it convenient for my friend - the way it's configured as already attached to the Oppo player:
Oppo BDP-105<-- Tesla T2 SE <-- PowerCell 4 <-- Tesla T3 SE <-- Wall outlet

I'll spend some time talking about the T2 SE since it's the cable directly connected to the Oppo. You can have a look at the manufacturer's information at their website if unfamiliar with this cable. Although not the "top of the line" AC cord, this unit has most of the "headline features" which supposedly provides benefits. It's got some kind of silver & copper conductor construction, "Tricon" and "T2" (?) geometry, high quality "G 07" IEC plugs... Then there's the "Quantum Tunneling" - some kind of 2 megaV pulsatile "treatment" that transforms "the entire cable at a molecular level" (what molecule(s) they did not say...). Check out some more pictures here.

Finally, we have the well advertised "Active Shielding". The claim here seems to be that using an electrically active (DC current) shield improves noise level and some how "greater frequency extension from top to bottom." We are of course not graced with any charts/graphs/details as to how this was determined.

To make things even more "enigmatic", we have these "Enigma Bullets" a.k.a. "Active Shielding Modules" capable of "tuning" the sound! They screw into the pigtails hanging off the male ends of the power cables. Silver = "open and airy", Black = "warm and rich", Grey = kinda in between. Hopefully the pictures below clarify the description:
Despite the huge calibre of the T2 SE cable, it actually feels rather hollow so it's hard to tell what wire gauge is being used inside.

Here they are, the "Bullets":
Cute, solid-feeling metal pieces with an electrical connector on one end and a little LED (the blue light shown in the 1st picture) on the other. They get warm plugged in to the DC source so I whipped out the multimeter and got 1400-ohms resistance for the black one, and curiously 1200-ohms for BOTH the silver and gray ones - not sure if this is supposed to be the case.

The "DC power" end of the "Active Shielding" is connected to what basically is a wallwart ("Mini Power Coupler") - here's a picture of them (2 for T2 SE, 1 for T3 SE) connected to the power bar:
Notice the $399 MSRP Synergistic "Quantum Line Strip" QLS-6 - power bar, no surge suppression - also "Quantum Tunneled"!

Of interest, you'll notice that one of the wallwarts had a sticker that fell off over time! Here's a close up of the label underneath:
A basic wall plug AC adaptor, 24V 300mA switching power supply you can order bulk from ENG Electric in China or Taiwan.

Here's the manufacturer's page on the PowerCell 4. It functions as a 4-outlet power bar. I don't think there's any surge suppression on it. It felt surprisingly light weight to me. I have no idea what they mean by a "magnetic cell" or what benefit that affords (?are there magnets in there?). Also, the comment about "The PowerCell 4 also improves picture quality on any display, with darker black levels, better color saturation and a more 3 dimensional picture, simply amazing" should be objectively assessable.

Finally, between the wall outlet and PowerCell 4 is the hefty Tesla T3 SE power cable. Again, here's the manufacturer information page. Looks to me that the main difference is the higher number of conductors (ie. thicker overall effective wire gauge) compared to the T2 SE above. You can read more about this PowerCell & T3 SE combination in this subjective review.
T3 SE cable plugged into wall. Notice its own lit "Enigma Bullet" and you can see the "Active Shielding" winding wrapped around the main power cable.
I could not find a price list all in one place... But the MSRP is something like this as of January 2014:
- Tesla T2 SE cable - $650/5ft (here)
- PowerCell 4 (North American) - $1,250 (here)
- Tesla T3 SE cable - $900/5ft (here)
----- Total MSRP for the set = $2800 USD

 

II. The Test

My friend lives in a multilevel condo and I figured that if indeed an expensive AC cable system is capable of cleaning up the noise coming through the outlet, then this is the kind of environment to demonstrate an advantage!

Of course, the comparison must be to a generic IEC cable, but lets make the generic AC cord even more disadvantaged - I'm going to add a 12' length of inexpensive extension cord to it. Measurements will be taken off the RCA output from an Oppo BDP-105 which he uses (the Oppo is an excellent USB DAC based on previous tests a year ago). This also gives me an opportunity to show a few measurements beyond my usual ASUS Essence One / Transporter / TEAC UD-501 trio of DACs.

Here then are your test "subjects":

A. Synergistic Research Tesla T2 SE to PowerCell 4 to Tesla T3 SE plugged into the condo wall plug:

B. Generic 6' 18AWG IEC AC cable I got 'free' in the box with something (black) + Generic 12' extension cable (white) into condo wall plug:


Test Setup:
Win 8.1 i5 Ultrabook --> shielded USB --> Oppo BDP-105 (powered with either A or B above into outlet) --> shielded (Tributaries) RCA --> EMU 0404USB --> shielded USB --> Win8 AMD X4 measurement laptop

- Newest Oppo USB driver (1.61)
- Latest RightMark Audio Analyzer (6.3.0)

 

III. Results

As usual, I'll measure at 16/44 to make sure the results cover standard CD-quality output. Then I measured 24/96 to get an idea of "high-res" performance. The tests were run under 3 conditions: Synergistic system without Active Shielding (wallwarts unplugged), Synergistic system with Active Shielding using Gray "Bullets", and finally the generic IEC cable + extender. (Note that in the labels I used "T2SE" but in fact the whole Synergistic chain was measured including PowerCell 4 and T3SE.)

16/44 (standard CD resolution):

Frequency Response
Noise Level
IMD + N
24/96 (high-resolution):
Frequency Response
Noise Level
IMD + N
As you can see... The expensive Synergistic power cords + PowerCell made absolutely no difference to the Oppo's analogue output compared to an absolutely generic power cable attached to extension cable for both standard resolution 16/44 and hi-res 24/96 test signals. Inter-test results were essentially exactly the same in all 3 conditions. No evidence here that the Active Shielding made any difference either.

For those who might wonder about jitter...

Again, no different. (Of course one cannot expect a power cable to affect jitter nor J-Test to be too anomalous through an asynchronous USB DAC.)

IV. Conclusion

Within the limits of the testing equipment - the EMU USB0404 as ADC - there is no difference using the Synergistic power cords with the Oppo BDP-105 compared to a generic 18AWG IEC power cable with extension cable in a multilevel condo building near the heart of the city.

Although the USB0404 isn't to be used professionally as test gear, as previously shown, it is a capable "measurement" device able to demonstrate very tiny effects like the -90.3dB LSB test, effect of digital filters, and slight differences between similar SPDIF digital transports; all of which I believe would be below the threshold of hearing for the vast majority of people. As such, I do believe the results above to be accurate and reflect reality when it indicates there is zero difference.

Could these fancy cables improve the sound from other devices like power amps or older technology like tube gear? I don't know... Remember though that tube equipment have much higher noise floor in general so even if this cable could lower it, the difference would likely be irrelevant. As usual, if power cables could substantially improve sound quality, why has there not been good evidence after all these years? I've often wondered why cables like these are not subjected to objective measurements like speakers, DACs, pre-amps, etc. in magazines like Stereophile using their fancy measurement devices? (Heck, many of these cables cost substantially more than good components!) Furthermore, right on Synergistic's web page, we are told that the "Active Shielding" lowers conventional parameters like noise floor and frequency response ("this closed circuit design not only improved subjective performance, but also made our cables measureably (sic) quieter, thus improving detail with greater frequency extension from top to bottom..."). So where are those measurements, and under what conditions? Peripherally, gimmicky marketing terms like "Quantum Tunneling" as it refers to the process they use really should be better explained (seriously, any time a company starts referring to Quantum-anything in the macroscopic world, it's best to be cautious). Finally, if sonic improvements can be made with a cable, did Oppo not bundle one they've tested to be optimal despite all their other engineering efforts?

Subjectively, I have heard my friend's system with the Synergistic cabling and with the generic power cable (as well as other systems with fancy cables but not to this degree of testing). To be honest, there's really not much to say subjectively with any certainty doing a tedious A/B/A trial. Do I think the Synergistics make the sound better? I'll go with the objective results and say this is most unlikely... Without a special setup, it's essentially impossible to do an accurate A/B comparison since there would be too much delay between cable switches, Oppo boot up, then start playing a song to really make any reliable comparison based on auditory memory of mental markers for high-fidelity. If audio qualitative differences were big enough, of course this could be a trivial task, but for at best tiny differences as in this case (if any), I do not believe this is possible based on research into the limitations of echoic memory. Good to see the measurements at least did not show any worsening using the exotic cables. All I can factually say is that the Oppo BDP-105 sounds great (and measurements demonstrate this high fidelity) through his system irrespective of which power cable(s)!

When it comes to power cables in my home system, my internal wiring is standard 14AWG copper and that's the "best" it's ever going to get in terms of power distribution. I don't see how passive wires can do anything for noise floor or "control resonance" or such beliefs. I'm quite happy with generic shielded 18AWG IEC cables for low power devices like the DAC, pre-amp, DSP equalizer. For higher power devices like the monoblock Emotiva XPA-1L and Onkyo AV receiver I have 16AWG shielded generic cables. I do not believe I hear a difference between the 18 and 16 gauge cables with the amps (in fact at one point I had an 18AWG cable on one monoblock and 16AWG the other and notice no stereo imbalance, noise, etc... even at high volumes) but I guess at least it makes me feel good that I did a little more to feed the neurosis :-).

As usual, please feel free to drop a link if you come across other tests on cables such as these; especially tests which have shown significant differences.

Recommendations:

- I've been listening to Eric Bibb's Blues, Ballads & Work Songs (2011) recently and I'm enjoying this Opus 3 SACD (was listening to the PCM layer on my friend's system the night of the Synergistic testing in fact). Easily accessible and great resolution!

- Mark Waldrep (aka "Dr. AIX") runs a nice blog at Real HD-Audio. Opinions and insights from a respected figure in the high-fidelity/audiophile world who clearly "keeps it real". If you haven't, I would highly recommend having a listen to some of AIX's recordings; especially in multichannel and the samplers provide a taste of his work. He posted an amusing recent anecdote on the use of a standard 75-ohm cable for digital audio at CES 2014. Also calling out the snakeoil on these "treatment" products sold through Blue Coast Records  - I wonder if they work better in PCM vs. DSD :-) (Dr. AIX's blog post). Respect.

Enjoy the music... And keep it real, folks :-).

PS: A big thanks to my friend for offering and helping with the testing - he has of course reviewed this write up for accuracy.


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

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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?)


-------------------------------------------------------
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.)

MUSINGS: Saturday AM B&W Nautilus Listening

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

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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...

-----

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.

MEASUREMENTS: Belkin PureAV PF60 Power Conditioner.

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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.

----------
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 Branford Marsalis tomorrow night... Enjoy the music everyone.

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

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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):

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 Discoveryfrom 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.
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?

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

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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"?

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