MEASUREMENTS: AUNE X1 (Mark I).

February 28, 2013, 1:35 pm

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As we all know, manufacturing capacity has shifted to Asia over the decades. Almost all the high tech gear we use in our daily lives are "Made in China" or Korea, or Japan these days. So why not commoditization of high performance audio and buying "the good stuff" from there as well?

Over the last decade, there has been a nice proliferation of extremely high performing and inexpensive gear coming out of Asia available on places like eBay among other on-line merchants. Last year, I was scouring the 'Net looking for a decent headphone amp/DAC combination when I came across some good reviews of the AUNE X1.



The USB implementation is the bog standard BB PCM2707 (adaptive isochronous, only up to 16/48 sampling rate USB 1). The internal DAC is the BB PCM1793.

Test setup:
Adaptive USB: i7 PC --> shielded 3' USB cable --> AUNE X1 --> shielded 3' RCA --> E-MU 0404USB
TosLink & Coaxial: SB Touch (EDO plugin, WiFi) --> standard coaxial/TosLink --> AUNE X1 --> shielded 3' RCA --> E-MU 0404USB

Supposedly, these adaptive USB DAC's are prone to jitter (I'll consider more about this in a followup post), but let's have a look at the analogue output results for now:


Already this looks good for RCA output. Despite the asynchronous design, the 16/44 USB path measures a little better than going through the coaxial or TosLink SPDIF from the Squeezebox Touch. However, due to the limitation of the old USB1 interface, it is not capable of hi-res 24-bit data or higher sampling rates.

Nice to see that the 24/192 sample rate result looks very good, essentially not giving up anything measurable compared to 24/96 using the E-MU 0404USB.

Frequency response:
16/44:

 24/96:

24/192 (coaxial only):

Nice.

Noise floor: (for 24/96)

Clean.

THD: (for 24/96)

TosLink just a tad noisier in the graphs above.

Conclusion:
These are just the measurements of the analogue output from the AUNE X1 as a line-level DAC. It looks very good and can serve as an upgrade to the standard Touch analogue output with about an extra 4dB improvement in the noise floor and dynamic range with 24-bit music. It loses a little bit in the 16/44 performance for noise level but makes up for it in a pretty big way with improved stereo crosstalk (subjectively it sounds better through headphones as well)...

As I mentioned above, I bought this unit as a headphone amp and DAC which I've been listening to this past year using my MacBook Pro TosLink output while doing work at night. It sounds very good - tight bass, nice stereo separation, very detailed with my Sennheiser HD800 - and can easily drive the more demanding AKG Q701 headphones to headache-inducing levels. I've hooked it up to a bookshelf system to make sure the RCA out sounded good (it does as suggested by these measurements), but otherwise have only listened to them through the headphones. Subjectively I have not been able to tell a difference between the same 16/44 track played back using the adaptive USB vs. TosLink vs. coaxial.

These days I see there is a Mark II with upgraded USB input using the Tenor chipset (up to 24/96) which should be an improvement. This is the level of performance you get for $200USD on-line. Commoditization of hi-fi is a very good thing.

In the next post, I'll have a look at jitter through this unit among other tests...

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MEASUREMENTS: Transporter TosLink vs. AES/EBU Behringer DEQ2496 Loopback.

March 6, 2013, 8:51 am

≫ Next: MEASUREMENTS: The hunt for load-induced jitter...

≪ Previous: MEASUREMENTS: AUNE X1 (Mark I).

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A little while ago, I demonstrated that the TosLink loopback with the Behringer DEQ2496 in line worsened the Transporter's jitter measurements significantly here.  Although I do not believe the extra 2ns or so of jitter was audible, I wondered if using an alternate interface than TosLink would have improved the situation. Although the Behringer doesn't have a coaxial SPDIF, it does feature the AES/EBU interface which is a digital balanced cable for me to try.

So, I got a couple of  5' Mogami W3080 + Neutrik connector cables for total ~$65 shipped to take the AES/EBU interface for a spin.

Setup: Transporter --> AES/EBU --> Behringer DEQ2496 (bypass mode) --> AES/EBU --> Transporter --> shielded 6' RCA (Tributaries brand) --> E-MU 0404USB --> AMD X4 laptop (Win8)

For the TosLink condition, I'm using 2x3' generic plastic TosLink cables instead of the AES/EBU.

Firstly, I wanted to make sure the analogue output remains good/unchanged:

The upper table contains 24/88 measurements - these days, it seems more hi-res is available in 24/88 (often SACD/DSD conversions), I figured it would be good to have a look to make sure it all measured well. The lower chart are the same conditions at 24/96. Note that these are the RCA output measurements so a little lower than with the XLR results posted before. Note that there is some inter-test variability compared to my previous results with the Transporter but generally we're talking <1dB difference.

As you can see, from the analogue perspective, there isn't any difference whether I'm measuring the Transporter DAC directly, or if it's running through a total of 6' TosLink or 10' AES/EBU through the Behringer DEQ2496.

Let us turn our attention to those pesky jitters. As usual, using the Dunn J-Test:

Transporter direct (16/44):

 

Transporter-DEQ2496 AES/EBU loopback (16/44):

Transporter-DEQ2496 TosLink loopback (16/44):

At 16/44, the Transporter is very clean for both direct and AES/EBU loopback. Most of these spikes are just J-Test modulation products  from the 16-bit signal being recorded in 24-bits. As you can see, the TosLink loopback is considerably worse with more sidebands congregated around the primary 11kHz signal. You can also see that using the TosLink interface with the Transporter slightly raises the absolute noise floor in general.

The AES/EBU loopback does add a small amount of jitter to the graph but it really is quite insignificant!

In the 24-bit domain, here are the same conditions with the 24/48 J-Test:

Transporter direct (24/48):

 Transporter-DEQ2496 AES/EBU loopback (24/48):

Transporter-DEQ2496 TosLink loopback (24/48):

These graphs look a bit different from the previous Transporter jitter measurements because I'm measuring RCA output (rather than the XLR from before which has a lower noise floor). Nonetheless, the results are the same in terms of jitter - TosLink is significantly worse.

Conclusion:

Well, I think that's it for my suite of Transporter measurements... The practical side realizes that jitter at these levels even with TosLink doesn't make an iota of difference playing real music. However, the audiophile seeks for "perfection" in as much as it's possible. Using AES/EBU digital cables instead of TosLink in this loopback configuration with the Behringer DEQ2496 for room-EQ does not add any significant extra jitter within the resolution limits of my test equipment. Cool.

MEASUREMENTS: The hunt for load-induced jitter...

March 7, 2013, 10:52 pm

≫ Next: MEASUREMENTS: Pioneer DV-588A - DVD-A and SACD "universal" player!

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I can't help it, I guess... I remain fascinated by this whole jitter issue, particularly with the belief that computer load can somehow alter the jitter/timing characteristic of a digital interface. After years of talk in the press about jitter, there's almost a mystique about this phenomenon. After all, jitter seems to be the central tenet to the concept that "bits are not just bits" because there is a timing component which somehow gets altered by the transport of said bits into the DAC for playback...  As I have alluded in previous posts, software-side techniques to reduce timing errors have been reported to include:
- strip down OS'es - prevent unnecessary processes from running
- prefer simpler/older OS'es like Windows XP, or alternate OS like Linux with RT kernels
- use large memory buffers
- specialized music player software which presumably go beyond bit perfect
- play only WAV / AIFF files (ie. FLAC decoding can add jitter)

Closely related hardware mods include:
- use SSD instead of HD
- slower/faster RAM
- slower/faster CPU & FSB

While I would not be able to test out all these assertions, I think we can logically deduce that IF timing is an issue and can be altered by computer load (whether due to OS, player software, FLAC decode, etc...), we should see some kind of anomaly with the Dunn J-Test when the computer gets busy in realtime.

Here's a video showing realtime analysis of my computer's motherboard TosLink audio sent to the AUNE X1 DAC.

Setup: i7-3770K + ASrock Z77 Extreme4 TosLink (Win 8, no special tweaks) --> standard plastic TosLink --> AUNE X1 DAC --> shielded RCA --> E-MU 0404USB

(Notice a few hiccups with the realtime spectrum analyzer... Quite alot of numbers to be crunched to plot out the 131,072 point FFT while recording video and audio.)

Realize just how "basic" this setup is... I'm using just the built-in motherboard TosLink straight to the $200 DAC.

As I have shown so far in the other tests, jitter is measurable between different interfaces. Also, electrical noise is easily measurable (for example, see the "NOISY i7" condition with the Essence One testing). However, I am still unable to show that multitasking or running the CPU at high load has any ability to change the timing/jitter characteristics of the Dunn J-Test significantly much less to an audible degree. As far as I can tell, the jitter phenomenon is a property of the digital hardware interface itself (ie. TosLink and adaptive USB tend to be worse than coaxial, AES/EBU, and asynchronous USB). Therefore, my suspicion/belief is that so long as the computer software player is functioning properly (ie. no buffer under runs, feeding a bitperfect driver like ASIO), then there should not be any jitter issues other than the limitation of the computer-to-DAC interface (at least in this case with a modern CPU & motherboard chipset).

In an even more extreme situation, with the AUNE X1 adaptive USB interface running off a USB hub with a hard drive attached transferring 25MB/sec data plus the CPU running 100% while playing the jitter test, I have not seen deterioration in the J-Test spectrum. (I won't bore you with the J-Test graphs since they look exactly the same whether computer is idle or busy and transferring files over the USB hub!)

Realize that this finding is very good! It means that we're free to do stuff like realtime transcoding and use of fancy upsampling algorithms without fear that somehow it will deteriorate the sound and that jitter is an independent variable not affected by the audio processing itself.

If anyone has reason to believe otherwise, I'd love to have a look at the test set-up and evidence of software-induced jitter (especially if it's audible!).

--------------------
Addenda:

Since I'm obsessive-compulsive and pedantic, here are a few measurements related to the above...

1. Just to show what a modern motherboard's analogue output looks like (ASrock Z77 Extreme4 motherboard "HD" sound, RealTek ALC898 codec, all "enhancements" off). Here are the measurements using a shielded phono-to-RCA cable to the E-MU 0404USB:

24/96 Frequency Response:

24/96 THD graph:

Summary - OK for 16-bit audio in terms of noise floor and dynamic range. Incapable of going beyond 16-bit dynamic range at best with 24-bit audio data. I suspect this is quite typical of on-board sound output. Notice the deterioration with 24/192.

2. Does the jitter pattern from the motherboard's analogue output itself get worse with increased CPU load?

Setup: Analogue output from ASrock Z77 Extreme4 --> shielded Phono to RCA --> E-MU 0404USB
Used the 24/48 J-Test like in the video above (that one was of course with the computer TosLink).

Computer idle:

Computer with 6 of 8 threads running at 100%:

Computer with 6 threads running at 100% + GPU (nVidia GTX 570) at 100% (FurMark running):

Conclusion: The motherboard's internal DAC is relatively jittery compared to the USB DAC's tested below. But symmetrical jitter sidebands are no different whether CPU or GPU load high. Bottom line...  I can't even seem to stimulate more jitter with the motherboard's own internal DAC by increasing CPU or GPU load. One consistent finding though is that bit of noise between 8-9kHz - usually whenever I strain the GPU.

3. I alluded to the AUNE X1 adaptive USB interface put under stress. Remember that in this case I'm using a separate external USB hub (not even the direct motherboard USB port); here are the boring graphs - 16-bit J-Test since the adaptive interface is incapable of 24-bit:

Computer idle: (note the 15kHz distortion at -90dB - discovered this to be a driver issue with ASIO4All - see April 1 update... Might never have noticed this if not for testing.)

Computer with 6 of 8 threads running 100%:

Computer with 6 threads running 100% + USB HD copying files at 25MB/s running off the external USB hub!

No difference... Yawn... Can't even get the adaptive USB DAC to show me bad jitter under these kinds of USB conditions!

4.Asynchronous C-Media CM6631A USB --> TosLink --> AUNE X1, same conditions as above plugged into external USB hub (16-bit J-Test for consistency):

Computer idle:

 Computer with 6 of 8 threads running 100%:

Computer with 6 threads running 100% + USB HD copying files at 25MB/s running off the external USB hub!

No difference... Very good graphs with minimal jitter (the spikes are primarily the 16-bit jitter modulation signal from the J-Test). Clearly the asynchronous interface is better even going through the TosLink.

This whole 'jitter' thing is getting tired and boring :-). Yawn...  I'm not using any exotic or expensive gear at all and yet I can't even get jitter anomalies to show up despite the strain I've put on the CPU/GPU and USB interface.

MEASUREMENTS: Pioneer DV-588A - DVD-A and SACD "universal" player!

March 11, 2013, 11:08 pm

≫ Next: MEASUREMENTS: Sony Playstation 1 (PS1) - SCPH-5501 as CD player.

≪ Previous: MEASUREMENTS: The hunt for load-induced jitter...

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Hey guys, remember the SACD vs. DVD-A "war" back in the early-mid 2000's?

In the heat of the battle, it was nice that a few manufacturers gave us these "universal" DVD players that could handle both competing formats. Pioneer was one of them and in mid-2003, released to the world the very reasonably priced DV-563A (~$200). A year or so later, this model under consideration, the DV-588A was released. I remember browsing around Future Shop (owned by BestBuy these days) in 2005 to pick up this unit since my previous Panasonic DVD player failed on me. At <$200, I figured I couldn't go wrong since this would 'in a pinch' also play my small collection of DVD-A and larger collection of SACD's.

The innards of this player revolves around the MediaTek MT1389EE SoC chipset which graced many <$400 players back in the day (including the Oppo DV-981HD and Sony NS955 [chip apparently relabeled as Sony CXD9804R]). Although it has been said that this chip is capable of "PureDSD", I do not believe any of the lower priced units operated in this mode because multichannel and bass management were performed in the PCM domain. As a result, the 1-bit DSD (actually DSD64 for the 64 x 44.1kHz = 2.8224MHz SACD sampling rate) in these machines are converted to 24/88 prior to conversion by the DAC (supposedly a BB DAC is used in this unit).

With the advent of PS3 SACD digital ripping in 2011 and a firmware upgrade to the Pioneer, this unit joined the ranks of the relatively few SACD players capable of playing backup SACD-R's.

Given the low price, let's see how well this player measures... First I created a DVD-A with the RightMark test signals along with the J-Test to have a look at how it performs as a hi-res PCM unit.

Setup: Pioneer DV-588A --> 6' shielded RCA --> E-MU 0404USB --> AMD X4 laptop, battery power, Win 8 x64
- Equipment plugged into Belkin PureAV PF60 power filter (same general setup as what I used for the Transporter previously).

Okay, so far not too bad for a budget player...  It can benefit from 24-bit audio with a dynamic range around 17-bits. Here's the 24/192 frequency response:

And here's the noise level (useful when we look at the SACD graph) - reasonably flat to about 50kHz:

As usual, I plotted the Dunn Jitter Test (24/48):

Quite jittery as you can see with a number of sidebands congregated +/- 1kHz around the 12kHz main signal.

Now as I mentioned above, this player is capable of SACD-R playback. So, I downloaded KORG AudioGate and went about converting the 24/192 PCM RightMark test and calibration signals and 24/48 Dunn J-Test into DSD64. With the help of a friend who's into this stuff, we got the tracks authored onto a SACD-R for testing.

Here's what RightMark looks like playing the SACD-R:

Not good... Just marginally better than DVD-A 16/44.

Lets have a look at the frequency response curve:

Not unexpectedly, the fact that it's resampled to 24/88 leads to an earlier cut off in frequency response than the DVD-A 24/192 spectrum above. I wonder if AudioGate is imposing a low pass filter starting around 30kHz to create that early steep drop off...

Noise spectrum:

Note that extra "bump" at 25 to 44kHz thanks to the DSD noise shaping (absent in the PCM noise graph above) - remember that this is a log scale so it's all scrunched up in the corner. The fact that the noise gets truncated at ~44kHz makes sense for the 24/88 PCM conversion, that's why I wondered above if AudioGate is cutting off the frequencies of the test signal earlier at 30kHz.

Although the Dunn J-Test is invalid in terms of stimulating worse-case jitter in the DSD domain, for the heck of it here's the spectrum (again 24/48 test):

It's cleaner than the PCM graph above. Possibly the resampling from DSD to PCM done internally - like ASRC's - is cleaner than the direct PCM from the DVD transport (I don't think it has to do with corruption of the LSB jitter modulation tone since even a straight 12kHz sine wave looks very jittery in DVD-A).

Summary:
Hey, I got to measure a DVD-A / SACD player with my custom SACD-R & DVD-A disks :-)! I've often wondered how this old Pioneer compared in terms of SACD vs. DVD-A playback. Subjectively, I've always thought it sounded good but not exceptional. I never bought the same title on both DVD-A and SACD to actually compare how the two would sound. Even if I did, there would be no assurance that the mastering would be identical anyways. Although I have about 30 SACD's still, they're barely listened to compared to the computer music server and my Squeezebox players. The fact that I do have a dedicated SACD changer still (the venerable Sony SCD-CD775) also means I don't bother with the Pioneer for music playback the few times I spin a disk not for the purpose of audio ripping.

Overall, the objective data points to a 'fair' DVD-A player for its price and a 'mediocre' SACD player with dynamic range just a hair above that of an ideal 16-bit CD. Of course I'm converting my test tracks from PCM to DSD and then the player is re-converting back to PCM again so this will affect the sound quality negatively...  Furthermore, I haven't played with AudioGate enough to get a sense of how good it is as a PCM to DSD converter compared to something like Weiss Saracon which is much too expensive for the casual hobbyist. Nonetheless, since RightMark is measuring dynamic range at 1kHz (should be quite good for DSD64), it should still measure better than what I got assuming the digital data conversion was done competently and the 24/88 internal conversion algorithm is good.

The question of course is how well does a good "pure DSD" SACD player measure compared to something like this budget Pioneer? Can anyone suggest a good SACD unit to try which can play back SACD-R's?

[It would have been nice to convert 24/176 test audio to DSD rather than 24/192 due to the even multiple 44kHz sampling rate. Unfortunately the EMU-0404USB seems unable to handle this sampling rate well for me. Ideally of course, DXD 24/352 would be even better!]

MEASUREMENTS: Sony Playstation 1 (PS1) - SCPH-5501 as CD player.

March 23, 2013, 12:44 pm

≫ Next: MEASUREMENTS: [UPDATE] Adaptive AUNE X1, Asynchronous "Breeze Audio" CM6631A USB, and Jitter...

≪ Previous: MEASUREMENTS: Pioneer DV-588A - DVD-A and SACD "universal" player!

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I got a kick out of this article by Stereophile awhile back (2008):
http://www.stereophile.com/cdplayers/708play/

Imagine, audiophiles using an "archaic" (circa 1994) game console as a CD player for thousands of dollars worth of expensive amps and speakers downstream!

That Stereophile article reviewed the first PS1 version which was the SCPH-1001. Instead of that older model, what I have here is the slightly later SCPH-5501; said to be superior for audio by some PS1 aficionados!

Here's an interesting post from a fellow on the Steve Hoffman message board going by the alias "rhing":

I recently purchased a used Playstation1 Model SCPH-5501 from a local video game store for about $25. I bought it based on the fact that it uses the same Asahi Kasei AK4309AVM DAC as Model SCPH-1001. The most obvious difference between Models SCPH-1001 and SCPH-5501 is that Model SCPH-1001 is the only Playstation 1 with RCA stereo audio output jacks. For some people, this is important to getting the ultimate performance from a Playstation 1. Model SCPH-5501 outputs stereo audio through a 12-pin Sony A/V Multiport jack that uses an A/V breakout cable with RCA jacks for Right Audio (red), Left Audio (white) and Composite Video (yellow). In addition to the RCA jacks on the rear, Model SCPH-1001 also has the same Sony A/V Multiport jack. The benefit to using the A/V Multiport jack is that there are no cheap NJM2100 op amps in the signal path between the DAC and the output. The RCA jacks are connected through a pair of op amp buffers that can adversely effect the sound quality. In fact, most Playstation 1 audiophile modders bypass the op amp buffers, or use the A/V Multiport to get the same effect of better sound. Other improvements were incorporated in the later Model SCPH-5501:

1) An improved Nichicon SMPS that generated less heat that could distort adjoining plastic components such as the lid and chassis that could lead to mistracking or binding of a spinning CD disk.

2) Positioned the laser assembly away from the power supply to reduce heat damage and RFI noise.

3) Implemented an auto-biasing feature for proper tracking. Model SCPH-1001 requires manual biasing of the laser circuitry to maintain tracking.

...

He goes on comparing the PS1 with other gear he has. Good read. I can't verify the comments but it certainly sounds well thought out and worth considering. For those wondering, the AK4309 is a 1-bit delta sigma "bit stream" DAC.

So, without further ado, here it is:

Notice the lack of RCA direct outputs at the back; there's a supplied multi-AV cable with RCA out instead.

   

Setup:

Playstation 1 --> stock multi-AV to RCA cable --> E-MU 0404USB --> shielded USB --> AMD Windows laptop

I used the stock PS1 power cable as per the photograph.

RightMark result:

As you can see, for convenience in comparison, I included the 16/44 results for a number of the other DAC/streamer units I measured over the last while.

Compared to the rest, the PS1 is clearly outmatched in terms of dynamic range. These measurements indicate that it's capable of around 15-bit dynamic range. At first, I thought it may be the fact that this is a CD player with all kinds of electronics in there perhaps lowering the dynamic range. However, when you have a look at the AK4309 datasheet, indeed the rated dynamic range for this part is 'only' 90dB.

Frequency response:

Some slight deviance from flat response curve up above 3kHz - unlikely to be noticeable through speakers and room interactions. Note the green plot for the Muse Mini TDA1543x4 for comparison to show what a typical NOS DAC measures like. Old school early 90's 1-bit delta sigma vs. NOS! :-)

THD:

Since the Transporter uses the newer generation AKM DAC (AK4396), here are the THD plots in comparison. Obviously, the Transporter is capable of measurably lower noise level with a cleaner graph notably above 10kHz.

In comparison, the TDA1543 (NOS DAC) measures much worse with numerous harmonics:

Jitter (16/44 Dunn J-Test):

Looks fine. Sidebands peak below -100dB from the primary 11kHz signal.

Conclusion:

There you go, the Sony Playstation 1 SCPH-5501 measured as a CD player. I can't say how it sounds compared to the "first-version" SCPH-1001 since I've never heard that model. Last night, for some late-night R&R, I listened with this unit through my AUNE X1 as headphone amp with Sennheiser HD800 headphones. Tunes on tap: Muddy Waters "My Home Is In The Delta" (downsampled Classic Records HDAD), Cat Stevens "Wild World" (2011 Analogue Productions), Nat King Cole "The Very Thought Of You" (2010 Analogue Productions), Akon "I Wanna Love You", eRa "The Mass", Joe Satriani "Crowd Chant", Jheena Lodwick "It's Now Or Never", Stephen Layton & Britten Sinfonia "For Unto Us A Child Is Born" from Handel's Messiah. They all sound great through the headphones. Nice details throughout, bass nice and deep on "I Wanna Love You", no accentuated sibilance with female vocals (that Jheena Lodwick track can be nasty), "The Mass" may have sounded a bit congested during the louder & more complex segments but really nothing to detract from enjoyment. I do not believe the measured 'limited' dynamic range negatively affected enjoyment at normal listening levels at all. Remember that ~90dB dynamic range is better than the majority of analogue sources.

Clearly compared to modern DAC's, the PS1 has inferior noise performance and concomitantly lower dynamic range. The other measurements like THD and IMD are respectable and jitter is not of concern. If we look at the TDA1543 DAC unit (DAC chip designed around the same era), the NOS unit has better dynamic range but with much more THD and IMD along with the typical high frequency roll-off at 44kHz sampling rate (of course you could feed the NOS upsampled 88/96kHz data to smooth this out). Between the two ultimately, it's one of subjective preference, especially for a "colored" DAC like the TDA1543 NOS (I've yet to measure a tube analogue output DAC). Personally, I'm of the "technically perfect" camp and would probably pick the PS1 over the TDA1543 even if a bit noisier - reasonably low level of noise like this is generally less objectionable than distortion.
 
I don't have any digital gear from the late 80's left, but by the mid-90's, "CD players" like this one sound and measure fine...

MEASUREMENTS: [UPDATE] Adaptive AUNE X1, Asynchronous "Breeze Audio" CM6631A USB, and Jitter...

April 1, 2013, 12:11 am

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In late February, I received an interesting USB2 --> SPDIF box from Asia. It's based on the C-Media CM6631A chipset (it says "Breeze Audio" on the metal case and "CM6631-V1.4" on the PCB) which communicates with an asynchronous USB2 protocol (recall that the Asus Essence One uses the CM6631).

Total price ~$50USD shipped.

Although it's said to be plug-and-play compatible with Mac OS X, I've run into a few snags which I won't discuss here - just be aware if you're planning to use this on the Mac (SEE ADDENDUM - ISSUE FIXED!)...  However, I'll restrict these measurements to the Windows 8 environment only where the custom drivers work well.

The idea I wanted to explore was whether using an asynchronous converter like this connected to the AUNE X1 would be better in terms of jitter than the adaptive USB port of the X1 itself. Also, how sensitive is jitter to computer load?

Note that beyond jitter, there are some other obvious reasons to do this... Firstly, using the CM6631A box allows hi-res TosLink/coaxial output up to 24/192 if you don't have a USB2 DAC. Secondly, it might be better to keep peripherals all running at high speed if you're going to be plugging this into a USB2 hub.

Lets take a look at the adaptive USB jitter spectrum directly from the AUNE X1 (remember, only up to 16 bits, so I'll be using the 16/44 Dunn J-Test signal) [UPDATED April 1, 2013 - SEE BELOW]:
(Setup: i7 computer --> shielded USB --> AUNE X1 --> shielded RCA --> E-MU 0404USB)

Numerous sidebands with spurious noise. I guess this is pretty typical of a very pedestrian adaptive USB1 interface. It doesn't "look" good but for the most part, these spikes are down below 100dB from the 11kHz primary tone.

Not only is adaptive USB said to be bad for jitter, but some believe it's especially prone to timing errors if you're multitasking...  Lets see what happens when the CPU is under 100% load (Prime 95, 8 threads), and to make it worse, lets also run the GPU (nVidia GTX 570) 100% with FurMark:

Essentially no difference to the spectrum whether the computer is fully loaded or not! You see a bit of noise showing up between 8-9kHz but it's all very low amplitude (again >100dB below the 11kHz peak).

Let's now put the CM6631A asynchronous USB --> SPDIF in the chain. Here's the setup now starting with a proper coaxial SPDIF 'digital' cable (Acoustic Research brand):
i7 computer --> shielded USB --> CM6631A --> shielded digital coaxial --> AUNE X1 --> shielded RCA --> E-MU 0404USB

This is a very nice 16/44 J-Test graph with minimal data correlated sidebands and most of the peaks are just due to the 16-bit J-Test modulation tone.

What happens with the CPU and GPU running at 100%?

No difference!

What about we use the TosLink output from the CM6631A instead?
(Setup: i7 computer --> shielded USB --> CM6631A --> decent plastic TosLink --> AUNE X1 --> shielded RCA --> E-MU 0404USB)

In this setup, the TosLink interface is almost exactly the same as coaxial (I'm actually impressed by this since TosLink often tends to be significantly worse than coaxial).

What about CPU & GPU running full tilt at 100% using TosLink?

Again - no difference! (Ignore that FurMark banner... Just got in the way of the screen capture.)

OK. We've also heard that a poor coaxial cable could be bad for jitter...  That is, if there is severe impedance mismatch between connectors (supposed to be 75ohms), the theory goes that there could be reflections which could damage the digital signal transitions. Furthermore, it is said that a short 3' cable is worse than a longer cable due to the transition time for these reflections.

What then would jitter look like if I replaced a proper shielded coaxial with cheap "freebie" 3' RCA connector that came with an old Pioneer DVD player (I used the red connector)?

Eh? That's all? Really no different than with the "proper" coaxial cable!

How about with CPU and GPU running 100% using the cheap RCA cable?

Again, that's all!?

Realize that in the analogue domain, I can measurably prove that the cheapo RCA is worse than the shielded proper coaxial digital cable based on noise floor differences. But in terms of jitter (digital phenomenon) which we're stimulating here with the Dunn J-Test, in this "real world" setup, I cannot show any significant difference despite the likely poor impedance match and 3' length! (Note that I'm not saying a cheap RCA cable is as good as proper coaxial in general, just that in this inexpensive setup, it works fine... I have measured in other situations like with my ASUS Essence One paired up to a Squeezebox Touch where a cheap RCA cable actually resulted in very severe jitter issues.)

Summary:

1. The asynchronous USB2 interface works to lower jitter! Having said this, I think we have to also recognize that apart from some noise spikes, the actual jitter sidebands with the adaptive USB interface wasn't terrible and likely not audible given how low in amplitude they were.

2. Whether the computer was running full-tilt at 100% or not, I could not demonstrate any change in the jitter characteristic! This applied to adaptive USB as well as asynchronous USB --> SPDIF. IMO, this makes me question the belief in "software-induced" jitter. I suppose if your computer is totally bogged down such that the audio buffer cannot be filled on time, you'll hear severe audio degradation. However, some people believe even light multitasking worsens sound quality, or the importance of shutting down background OS processes or running "stripped down" OS'es. By extension, some people swear by various playback programs having an effect on the audio quality (ie. Decibel vs. Amarra vs. JPlay vs. Audirvana vs. iTunes vs. foobar...) even if all that's promised is "bit perfect" output. It almost always comes back to "jitter" as the explanation some how. Really? Is there any objective proof for such beliefs or even a testable hypothesis?

As I have hinted in previous posts, even though I (obviously!) believe that jitter exists and is measurable, I do not believe it is audible with music unless extreme like maybe >10ns. It's also worth remembering that research into the threshold of jitter audibility decreases with higher frequencies; but this also correlates with human hearing losing sensitivity the higher up we go - especially as we get older (therefore, sensitivity to jitter should decrease with age). These days, well engineered "high fidelity" gear should not be even close to showing 10ns jitter. Likewise, software player programs should be capable of "bit-perfect" output without difficulty or cause timing errors.

Over the years, I have also tried many software players like Amarra, Decibel, JPlay, etc... Ultimately I've never been convinced of significant differences in sound between them (so long as EQ and plugins not active). Foobar + ASIO driver sounds great to me on the PC side and I use Decibel for the Mac simply because it's inexpensive and works well to switch sampling rates.

Bottom Line:Don't worry about jitter! It's more than likely inaudible in a modern computer system and with decent (not necessarily expensive) audio gear. I see no evidence that high CPU/GPU load makes any difference to jitter. Isolating your DAC from electrical noise polluting the analogue output seems much more important.

ADDEDDUM (April 1, 2013):

I found out why there was the 15kHz peak in the adaptive USB1 tests with the AUNE X1. Had to do with some old ASIO4All drivers I had installed on the i7 computer...  A good reminder of how easy it can be to mess up settings when using a PC for audio playback!

Anyhow, for completeness, here are some updated graphs of the X1 using adaptive USB measured off my Win8 AMD Phenom X4 laptop. Note that even though the graphs look different, the conclusions are the same - asynchronous USB2 is less jittery, and when the computer is at high load, the noise floor worsens for the adaptive USB case only.

Adaptive USB1, AUNE X1 using WASAPI for 16/44 J-Test:

Now with 4 threads running Prime95:

Notice that slight noise floor elevation from 8-9kHz. Not as extreme as my i7 testbed with the powerful GPU running.

Asynchronous CM6631A --> shielded coaxial --> AUNE X1, WASAPI, 16/44 J-Test:

Nice and clean...  Looks just like with the i7 testbed. What happens with Prime95 running?

No change.

Asynchronous CM6631A --> TosLink --> AUNE X1, WASAPI, 16/44 J-Test:

Now again with Prim95 running on all 4 cores:

No difference...

OK. So I have now shown that the jitter results with the CM6631A device is reproducible on 2 platforms (i7 & AMD Phenom X4) as being low for both coaxial and TosLink. Also, the CM6631A device seems quite immune to noise from the main computer (compared to the adaptive USB interface). Both ASIO and WASAPI work well for the CM6641A. Still no evidence that the jitter pattern itself changes with increased CPU load.

ADDENDUM (March 31, 2013):

Noticed this very useful post on diyaudio.com from 'bvs':

I've recently found a solution for issue that caused when CM6631A module is connected to any Mac USB 2.0 ports. When you connect module to Mac it is detected as USB Audio Class 1.0 device and you have no ability to use anything more than 48/16. Module has a reset chip LM810M3-2.93 (http://www.nscrus.ru/content/catalog/pdf/LM810.pdf), but as we can see from the CM6631/32 datasheet, the CM6631A has a power-on self-reset. So, reset chip has been removed from a board and now the module is always correctly detected as USB Audio Class 2.0 device on any Mac USB port. This method has been tested on MacBook Pro, Mac Mini and this CM6631A module: New CM6631 USB Module Assembled Board for DAC3 AD1955 DAC7 WM8741 by Weiliang | eBay LM810M3-2.93 is a 3-pin chip located on the top of CM6631A.
So, with a needle-nosed plier, I went to work on removing that little 3-pin chip (labelled "SA B" on the board)...  Here's the result (chip removed): The unit works as expected on my MacBook Pro's now with full playback up to 24/192, no need for an external power supply, and no need for custom drivers. Remember, this modification is for the CM6631A only. Earth to Microsoft - isn't it about time we got native UAC2 driver support in the OS!? Especially considering that it's been available for OS X and Linux since 2010!

MEASUREMENTS: Oppo BDP-105 does DSD.

April 6, 2013, 12:37 pm

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Well, it's out...  The March 26, 2013 BETA firmware for the BDP-105 that allows native DSD playback from this unit's USB ports as DFF and DSF files on a USB stick. As far as I know, there are no plans currently to allow computer playback connected to the USB port as a DSD DAC.

With this recent development, I headed back to Phil's place to run a few more tests...

Here's the basic premise of what I did...

Using the freely available KORG AudioGate 2.3.1, synthetic 24/192 test signals from RightMark 6.2.5 were converted over to DSD64 (2.8MHz sampling) and DSD128 (5.6MHz) for testing. We soon found out that the DSD128 files created could not be played back properly by the Oppo (it would play but the DSD128 files had timing issues - played too slow). Presumably there is a bug here somewhere with the beta firmware or the AudioGate converter. As such, I was only able to test DSD64 playback.

The first thing done was to go through the Oppo's settings menu making sure there were no volume settings, bass management, etc. active. I believe if any of these are turned on, the DSD will get converted to PCM.

The hardware setup is similar to what I did with the original BDP-105 tests (only difference being use of the front USB connector):
Patriot Rage XT USB2 memory stick 16GB with test files --> front USB port of BDP-105 --> shielded 6' RCA --> E-MU 0404USB --> Win8 AMD X4 laptop

RightMark results:

The first 3 columns are the tests done in PCM mode at various hi-res sampling rates. These essentially measure <1 dB different compared to my original tests using the Oppo's USB asynchronous DAC. Nice confirmation of inter-test reliability.

The last column is with the 24/192 test signal converted to DSD with the KORG software. As you can see, it's almost the same. Note however that RightMark is calculating these parameters just within the audible spectrum between 20Hz - 20kHz (AES17 standard).

Frequency Response:

First hint/reminder of the DSD effect. There's some high frequency noise breaking through up in the 70kHz range. Otherwise, the curves are relatively comparable with -5dB extension out to around 40kHz for DSD and 50kHz for PCM 24/192.

Noise:

Demonstration of the DSD noise shaping through the Oppo. From 20kHz onwards, the noise level rises quite remarkably as you can see. It's all ultrasonic of course so unlikely to cause an audible problem and would only matter if this creates any strain on your amp/speaker system or if nonlinearities cause distortion in the audible spectrum.

Although also not a problem, notice the noise floor from about 12kHz to 20kHz is not as flat with DSD.

THD:

Another view of the ultrasonic noise.

Jitter:
The J-Test cannot be used with DSD of course. This is just for completeness. I've already shown previously that jitter isn't an issue with the Oppo...  Here's just what the 24-bit Dunn J-Test looks like going through DSD transcoding.

If we compare it to the PCM:

Note the loss of the regular modulation pattern. Basically this is telling us that the LSB in the 24-bit signal has been affected and effectively dithered over by the conversion process to DSD.

Analogue Output:
Lets now have a look at what a 1kHz -6dB sample looks like after going through the DSD process. What I did here was record a few seconds of a pure 1kHz test tone in 24/192 to have a look at the waveform zoomed in.

PCM 24/192 FLAC played back:

DSD64 KORG transcoded DFF file played back:

The high frequency noise in the DSD signal can be seen (you might have to click on the images to get a good look). Not a big deal in that this is not audible but a reminder that DSD64 cannot reproduce a simple 1kHz sine wave as smoothly as that produced by the reconstruction filter in the PCM domain.

Impression:
As I had hoped when I wrote that piece on the Pioneer DV-588A last month, here are some results from a device that performs "pure DSD" decoding.

Within the audible spectrum, DSD64 produced by the KORG AudioGate software looks good. Standard measurements like dynamic range, noise floor, distortion are all looking great and reminds us of the high level of performance the Oppo BDP-105 is capable of. I was disappointed that I could not get the KORG-encoded DSD128 test signals to play properly. I don't know if this is due to the KORG software or the beta firmware. Maybe I'd have better luck with Weiss Saracon if I had access to this conversion software... Oh well, maybe next time :-)

As a reminder, all the tests I've shown were converted from the 24/192 PCM domain into DSD64 and therefore will be subject to the limitations of the conversion software and PCM source (note that at 24/192, this is not likely a technical issue).

An interesting observation; even though the encoded DSD128 could not play, free downloads of demo material from 2L worked just fine on the Oppo! They sounded great with a wonderful sense of space, timbre, and dynamics.

Bottom line: The Oppo did a great job with DSD playback just like it did with PCM. Limitations of DSD are clearly seen (ultrasonic noise pollution mainly). From a purely technical perspective, within the 20Hz-20kHz audio spectrum, there's really nothing to differentiate all these hi-res formats. However, if you include ultrasonic characteristics, PCM is definitely cleaner.

Given the frequency response curve demonstrated, this KORG DSD64 conversion + Oppo playback system can likely be encapsulated within the parameters of a good 24/88 system.

MUSINGS: On SACD & DSD audio...

April 6, 2013, 9:29 pm

≫ Next: MEASUREMENTS: Laptop Audio Survey - Apple MacBook Pros, Acer Aspire, ASUS Taichi...

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SONY multi-page spread (with hybrid SACD 'RS500' sampler!) - Rolling Stones magazine, December 2003.

Okay, now that I have posted a number of measurements over the last months, permit me a moment to play "Speakers Corner" and talk a little about the state of affairs around DSD now in early 2013...

Firstly, remember that DSD isn't anything new. It's basically a digital format based on Delta-Sigma modulation which has been used in consumer digital audio since the late-80's and early 90's - remember those old Sony 1-bit and Panasonic MASH players from around that era? It was said that as the CD patents were set to expire, Sony and Philips decided to create a new disk format for the 21st century - hence the birth of SACD around 2000 using the "1-bit" encoding method instead of multibit-PCM carried on DVD-like disks capable of higher data capacity (multi-bit ended up being DVD-A of course). Sound quality was said to be better because of things like "100kHz frequency response", better noise floor and in time, multichannel. Of course in creating this new standard, copy protection was job one and they did a great job. I remember in 2004 visiting China and seeing pirated DVD-A's which were easily "cracked" soon after release, but not so SACD's.

It wasn't until 2011 when digital "ripping" of SACD's became a reality with the PS3 techniques easily found on the Internet. This finally allowed hobbyists to have a good look at the digital data "under the hood". What was seen in many cases was unflattering... The controversy around Norah Jone's "Come Away With Me"back in 2004 having been just 16/44 converted to DSD for the SACD release was really the tip of the iceberg. In fact, many albums were given the same treatment.  These "SACD's" were nothing more than just 16/44 PCM transcoded with no possible sonic improvement whatsoever - in fact, since the conversion process is not perfect, it's probably preferable to just buy the CD and enjoy technically 'better' data and save the cost differential (unless the SACD has a multichannel mix which could represent some extra value).

In terms of my personal experience with SACD, I bought my first SACD around 2001 and over the years have heard a number of high end models including the first Sony SCD-1, 555ES, 777ES, couple of Marantz, a Denon or two. I remember spending a whole evening back in late 2001 with the just-released Telarc 1812 Overture on hybrid SACD going back and forth between the stereo SACD vs. CD layers on the Sony SCD-555ES with a good pair of headphones late into the night. Despite the hype, my impression was that any differences were at best subtle.

As we all know, commercially, SACD was a failure. Sony ended their push into SACD disks around 2006 and likewise, most SACD hardware available in their product line these days exist as part of Blu-Ray players (how many actually still do "pure" DSD is unclear). [I was reminded by Geoff on Audio Asylum that the audio-only Sony SCD-XA5400ES is still available!] Newer versions of the PlayStation 3 likewise have lost the ability to play SACD.

On the software side, it's good to still see regular releases of SACD usually in the classical and jazz genres with some audiophile MoFi, Audio Fidelity goodies every once awhile.

In the last year, it's interesting to note the increase in DSD-capable USB DAC's (witness Benchmark's DAC2, Chord, Mytek, EMM, dCS, Esoteric, MSB, Meitner, etc... See List). It seems to be part of the feature-set du jour; perhaps much like the upsampling feature with various filter settings which started a few years ago. The fact is that many 24-bit DAC chips themselves have had the capability for DSD decoding for years now, but it only appears recently that manufacturers are adding this to the driver set since the DoP standard - and in doing so ticking off another feature on the product brochure.

But does this make any sense? Lets examine a few factors...

1. Is there any music available in DSD? The answer is of course YES. Thousands of SACD titles exist encoded as DSD64 in fact. But at present there does not exist a way to download this stuff legally. Many SACD rips are already out in the wild but unless you own a copy of the SACD, it would not be legal to obtain such copies... This even is contentious depending on the copyright legislation in your country since to get at the data, copy protection mechanisms have been circumvented which may not be legal.

2. Is the DSD music of technically good quality? What I'm getting at here is the concept of provenance of the music - where did it come from? There are of course small music studios like AIX or 2L where the music is recorded to the highest standards, but the vast majority of SACD/DSD music exists as analogue conversions of decades-old music, or if it's a modern digital recording, almost universally processed through the PCM route. Indeed, after more than a decade since the SACD, processing techniques beyond basic splits, fades, volume adjustments remain PCM-based. There really is no reason this will ever change.

As I noted above with Norah Jones as an example, there are MANY recordings on SACD which look like they're from a 44kHz PCM-sampled source. Here are just a few that I've come across over the years that look suspicious - MFSL Dead Can Dance remasters, Sony's Brubeck "Time Out", Baby Face "The Day", Uriah Heep "Magic Night", Ryan Adams "Rock N Roll", Joe Satriani "Engines of Creation", Albert King "I'll Play The Blues for You", Yo-Yo Ma "Soul Of The Tango", Blue Oyster Club "Agent of Fortune", "The Phantom Of The Opera" soundtrack... As far as I am aware, there isn't any database out there to keep track of this (nor do/should most people care at this point given the marginalization of the format). The point is, SACD's are "polluted" by recordings of "inferior" quality, many of whom look like they originated as transcoded CD.

3. Does DSD sound better? This is where we can get really contentious. Technically, as you can see from my Oppo BDP-105 measurements, DSD64 doesn't seem to offer anything that 24/88 PCM cannot already achieve. Over the years, Sony and others have advertised it as offering extended frequency response to "100kHz" (check out the table in this brochure), flat square wave response, or "perfect" impulse response free of ringing. Clearly the "100kHz" frequency response is misleading given the noise shaping going on in the ultrasonic range. You can read about the square waves in more detail here at Craigman Digital. As for impulse response, sure, it's good when you sample MILLIONS of times a second, but has anyone shown that the the human auditory system is capable of perceiving this in well conducted studies by correlating perfection of impulse response with superior quality? (Feel free to drop a reference!)

Also, there are instances where the SACD layer and CD layer of the same hybrid disk have different masterings, thus making it impossible to compare quality of the underlying technology (an example is Joe Satriani's "Engines Of Creation" - CD layer dynamically compressed badly at DR7 vs. stereo SACD layer DR11.)

As much as "audiophiles" claim otherwise, there is still no scientific research to show the benefits of hi-res audio from properly dithered 16/44 "CD quality" audio much less differentiate hi-res DSD from PCM (have a look at the Wiki for a few references on this). Remember also that we are at this point more than a decade since wide availability of SACD titles!

4. File formats: DFF (Philips) and DSF (DSF). In terms of having DSD files on your computer for these DSD DAC's, there are currently 2 common file formats out there. DFF unfortunately cannot hold metadata tags - this is very unfortunate. It makes it tough to catalog music and IMO excludes its usefulness in music libraries where one has hundreds or thousands of albums. DSF at least is better in this regard.

Yes, I know multi-terabyte HD's are cheap. But this is still no reason not to maximize storage space IMO and minimize transfer time when doing file copy or backing up drives. DST compression (lossless compression sort of like FLAC for PCM) is only available for DFF... Looks like for now you have to choose between tagging vs. compression of DSD files; and it's been like this for years so don't hold your breath.

Speaking of file size, DSD64 is about 680KB/sec (and DSD128 twice that at 1.37MB/s). Compare this with stereo 24/96 PCM at 560KB/sec (24/192 at 1.13MB/s) and you really wonder whether the storage requirements make sense for what you get in audio quality; especially when you figure in the fact that FLAC or ALAC or APE can generally reduce the bit rate further by 30-40% easily AND still retain full tagging function.

5. Economic incentive. Other than small audiophile specialty labels, selling DSD files doesn't really make sense to most companies does it? Storage requirements are higher as above, more bandwidth is needed for transmission over the Internet. You'd be selling to a vastly smaller number of installed DSD-capable DAC's (compared to every CD/DVD/Blu-Ray player, every DAC, every computer). There's no special Digital Rights Management so there's no copy protection. Physical disks are on their way out and since SACD's can now be ripped, good luck getting any of the major music audio labels interested.

Okay, time to wrap up... From personal subjective experience as well as thinking about this over the last while, I really see no reason for DSD to take off. Don't get me wrong, I actually very much enjoy my collection of SACD's but then I also enjoy good CD's, DVD-A's, Blu-Ray's, FLAC's, and high bitrate MP3's. Whether audible with real music or not, DSD64/SACD is capable of better-than-CD dynamic range in the audible spectrum which I believe is the main (only?) benefit in terms of stereo playback.

Other than as a feature to hype up the new DAC's, there really doesn't seem to be anything materially new with DSD DAC's. Certainly no "disruptive" technology has been introduced by this feature. IMO DSD64 doesn't technically measure or sound better than 24/96 (or even 24/88), the de facto standard for studio work for any new project remains PCM, and other than small audio companies targeting a specialized group of music lovers (where perhaps they can meet a reasonable financial target), it doesn't make sense for any major studio to invest capital into this "new" audio delivery format which already has failed to make a significant impact in terms of proven superior audible quality or economic viability even when it had full SACD copy protection, major marketing push (free Sony and Telarc samplers & promos back in the day!), and weak competition (errr...  DVD-A?). Remember, DSD has to compete now with hi-res PCM in a very PCM-friendly world.

In 5 years, once the "second coming of DSD" hype dies down with these new DAC's, I suspect we'll be looking back at DSD much like we see it today - a niche format, owned and loved by a loyal group of audiophiles probably consisting of mostly acoustic jazz and classical music fans (where maintaining DSD purity from recording to playback could make sense). We'll see...

Whichever way this goes, enjoy the music :-)

(Update: May 12, 2013 - just wanted to reiterate something I noted at the end of my TEAC UD-501 DAC DSD evaluation...)

Before I end off, I just want to make a general comment & plea about the state of DSD computer audio now that I've got a chance to try it out.

The DFF and DSF file formats are inadequate. Compared to FLAC, APE, or ALAC, these DSD file formats feel geriatric! Seriously, PLEASE get the file format right.

Firstly, we need good tagging features - all the more important for DSD since much of the excellent material consists of classical music where it's important to document conductor, orchestra, composers, title, year of performance and composition, etc... Please let me be able to use something universal like the excellent Mp3tag to manage all my PCM and DSD files.

Secondly, a standard DSD file format NEEDS lossless compression. DSD is extremely compressible - using DST, I regularly see compression ratios >2.5:1 losslessly, getting up to 3:1 in some tracks. This becomes even more useful for DSD128 where the space savings are very substantial. By doing this, DSD64 can be compressed to file sizes overall smaller than 24/88 encoded with FLAC with equivalent (some would say better) sound quality... I'd certainly be happy with that! Lossless compression would also save file transfer times and cost of storage for the music producer, distributor, and of course consumer. Seriously, what other modern hi-resolution media format doesn't allow for at least lossless compression?

Over the months, I have heard DSD apologists talk about how you don't need compression or native tagging because "hard drives are cheap" and "JRiver can tag with its internal database". Sorry... That's not good enough. This is a foundational matter and will impact future generations of products, so it's important to get it done properly instead of rely on work-arounds.

Please guys, now that you've gotten together to define DoP and manufacturers to make these DAC's, lets get it done right with a standard DSD format that's fully functional, preferably "free" as in "open". Maybe some enterprising coder like Josh Coalson of FLAC fame can apply their expertise!

Thanks in advance. ;-)

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MEASUREMENTS: Laptop Audio Survey - Apple MacBook Pros, Acer Aspire, ASUS Taichi...

April 13, 2013, 10:42 pm

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Now that the asynchronous CM6631A USB to SPDIF device works well on the Mac with full access to sample rates after removing that reset chip, let's have a look at a few of the laptops I have around here - how the internal DAC's measure and compare this to the effect of disparate hardware on adaptive USB and asynchronous USB DAC's. First up, let me introduce you to the laptops I'll be testing:

1. Early-2008 17" MacBook Pro (MBP):

     CPU:  2.6GHz Core 2 Duo
     Memory: 6GB 667MHz DDR2
     Graphics: nVidia GeForce 8600M GT 512MB
     OS: Mac OS X 10.8.2
     HD: Corsair GT 240GB SATA SSD
This has been my work laptop for a few years until a couple months back. Essentially top-of-the-line laptop back in 2008 when I bought it. Still looks good and works well! I updated the hard drive to an SSD and upped the 4GB to 6GB about a year back.

2. Mid-2009 15" MacBook Pro (MBP):

     CPU: 2.26GHz Core 2 Duo
     Memory: 8GB 1333MHz DDR3
     Graphics: nVidia GeForce 9400M 256MB
     OS: Mac OS X 10.8.3
     HD: WD 640GB SATA
My wife's work laptop. One of the Apple UniBody construction machines. The only laptop in this roundup with a hard drive rather than SSD.

3. Acer Aspire 5552-7858 (17"):

This is the least expensive of the laptops; priced at <$600 when I bought it new in 2011. This is also the machine I've been using over the last months for all my "mobile" data collection such as the Oppo tests at my friend's house. Mostly used by my kids and guests if they want to look something up. SSD upgrade about 6 months back.

     CPU: 2.2GHz AMD Phenom II X4 N970

     Memory: 6GB 1333MHz DDR3

     Graphics: ATI Mobility HD4250

     OS: Windows 8 x64

     HD: OCZ Vertex2 120GB SATA SSD

4. ASUS Taichi 21 DH51 (11.6" dual screen):

My newest "ultrabook" laptop (bought February 2013). I know the reviews are mixed but IMO, this is a fantastic mobile laptop / tablet with a high resolution 1080P matte screen in laptop mode and glossy touchscreen as a tablet! Everything in one light and mobile package; fantastic to use in the local coffee shop to update one's audio blog :-). Great for work and although the battery life is at best average in power saving mode, good enough for me with light Office duties.

     CPU: Intel Core i5-3317U (1.7GHz)

     Memory: 4GB 1333MHz DDR3

     Graphics: Intel HD 4000

     OS: Windows 8 x64

     HD: 128GB SSD mSATA

Note that this unit doesn't have USB2 - tested with USB3 ports.

I. NATIVE DAC (RightMark Analogue output analysis):

First, let us have a look at the internal DAC's in these laptops. Measurements were done as follows:
Laptop headphone out --> shielded phono-RCA cable --> E-MU 0404USB --> Aspire AMD laptop

In order to try to ensure "bit-perfect" output, I used Decibel 1.2 on the Macs (turned off all volume adjustment, allowed Decibel to have exclusive access, buffered audio files to memory). On the Windows 8 side, playback was with foobar2000 1.2.4 (latest stable version) playing back with either WASAPI with no dithering or ASIO if driver available.

As you can see, each laptop demonstrated significantly different measurement results. Every one is capable of handling 24/96. The frequency response graph for each machine looks different, especially between the MacBook's vs. Windows machines (perhaps the Acer and ASUS use similar OEM DAC hardware?).

Not surprisingly, the ASUS has the highest noise floor of the 4 machines. It is an "ultrabook" so there's a lot of electronics packed into that case plus the headphone out jack is right underneath the screen hinge (instead of closer up front by the keyboard like the other machines).

I'm impressed by the MacBooks though with >100dB dynamic range! That's fantastic for laptops and certainly as good as lesser "audiophile" DAC's out there.

 II. Adaptive Isochronous USB (RightMark Analogue output analysis):

Using the AUNE X1 (Mark I) as the DAC, let's look at the relative performance of each machine using the X1's USB interface - the old "adaptive isochronous" BB PCM2707 USB1 technology measured at the usual CD sample rate of 16/44.

Test laptop --> Shielded USB cable --> AUNE X1 --> Shielded RCA --> E-MU 0404USB --> Aspire laptop

Notice how close the results are except for the Aspire-X1 results. Obviously since we're now utilizing the same DAC for each machine, the sound output is dictated by the properties of the DAC. The difference with the Aspire-X1 setup is that the Aspire is both the test machine and the measurement device (loopback setup). This means the Aspire needed more CPU processing and had much higher USB traffic than the other machines (maybe the mixed USB1 & USB2 modes have something to do with this). What we're seeing is likely electrical noise generated by the busy machine, and as I'll show later, this is NOT likely due to significant timing jitter in the digital domain.

Otherwise, I see no significant difference in measured "sound" coming out of the DAC whether the machine is Mac or Windows despite the significantly different underlying hardware, OS, playback software, etc... 

 III. Asynchronous USB (RightMark Analogue output analysis):

Test laptop --> shielded USB2 --> CM6631A --> inexpensive decent 3' TosLink --> AUNE X1 --> E-MU 0404USB --> Aspire laptop

Look at what happened with the asynchronous and very inexpensive ($50) CM6631A performing USB to SPDIF duties! Audio output from the AUNE X1 DAC is now IDENTICAL from every laptop. This shows that compared to the USB1 input to the X1, the CM6631A converter, in using TosLink provides excellent electrical noise rejection from the USB port (note that I'm running the converter USB powered; NOT separate power supply even). I think this is a nice demonstration of how the TosLink interface has its place despite measurably worse jitter in some cases.

IV. Jitter - Native DAC:

The Dunn J-Test was created to look at jitter in SPDIF interfaces, therefore there really should not be any issue with built-in DAC's for each laptop. Reminder, "bit-perfect" settings for Decibel with the Mac's, foobar2000 WASAPI 16-bit undithered for the Windows machines used. Here's how the 16/44 J-Test result looks for each machine:

2008 17" MacBook Pro:

2009 15" MacBook Pro:

Acer Aspire:

ASUS Taichi:

Quite different spectra as you can see. What is learned from the graphs is NOT jitter but rather the loss of the jitter modulation pattern with the 2008 MacBook, Acer Aspire, and ASUS Taichi. This means that despite using bit-perfect settings with the players, these computer DAC's are incapable of "bit-perfection". Other than the 2009 MacBook Pro, all the other machines either truncate or perhaps have some kind of 'forced' dithering in place such that the LSB jitter modulation has been corrupted.

V. Jitter - Adaptive Isochronous USB AUNE X1:

Using the same method, lets look at the jitter pattern with the USB1 interface built into the AUNE X1 (Mark I). Again either using Decibel or foobar2000 with bit-perfect settings in place. USB cable is generic shielded good quality 6' cable:

2008 MacBook Pro to AUNE X1:

2009 MacBook Pro to AUNE X1:

Acer Aspire to AUNE X1:

ASUS Taichi to AUNE X1:

As you can see, the jitter modulation tone can be see in each situation. This confirms that indeed bit-perfect data is being sent to the DAC containing the LSB data. If anything, it looks like the Mac's may have a few more jitter sidebands than the Windows machines. If you recall from the RightMark results above, the Acer Aspire is performing both playback and recording duties in this situation so you can see that the noise floor is higher than for the others (noise increases significantly in the lower frequencies as shown in the RightMark graphs) likely due to the increased CPU and USB activity causing noise to spill over into the X1 DAC through the USB electrical interface.

VI. Jitter - Asynchronous CM6631A USB to TosLink to AUNE X1:

What about when the CM6631A is used? Two things are happening here - USB communication is now asynchronous allowing better clocking (ie. lower jitter), and galvanic isolation is also happening by way of the TosLink SPDIF intermediary. As I measured in a previous blog post, the CM6631A appears to have just as low jitter whether with coaxial or TosLink.

2008 MBP to CM6631A to AUNE X1 (TosLink):

2009 MBP to CM6631A to AUNE X1 (TosLink):

Acer Aspire to CM6631A to AUNE X1 (TosLink):

ASUS Taichi to CM6631A to AUNE X1 (TosLink):

Notice how the jitter spectrum looks almost identical now just as the RightMark measurements above look essentially identical in each case with the CM6631A. With TosLink galvanic isolation, the Aspire's noise floor is in line with the others and in each case, the jitter modulation signal is well defined, again proving that Decibel (sending to the Mac's native USB 2 Audio driver) and foobar2000 (using the C-Media ASIO driver now) are capable of bit-perfect output when the hardware allows.

VII. SUMMARY:

1. Even though most laptops/motherboards these days advertise their audio hardware as "HD Audio", they are generally nothing of the sort. Although the MacBook Pro's showed good dynamic range in the RightMark tests at 24/96, the 2008 model was incapable of true 16-bit processing. Likewise, neither of the two Windows 8 laptops were capable of preserving the 16-bit LSB jitter modulation pattern. Perhaps obvious, but the bottom line is that you cannot trust the built-in DAC's for even 16/44 resolution audio.

2. Watch out for noise creeping into the DAC from the USB interface as shown with the Aspire-X1 Adaptive USB example. This can originate from internal electrical components as in this case or other times ground loops. Galvanic isolation with the TosLink interface is one way of dealing with this.

3. Overall, the asynchronous USB interface is better than adaptive isochronous USB for jitter in this example (obviously I cannot vouch for all asynchronous DAC / interfaces). This has been demonstrated before already and seen again here. These days it's a moot point since asynchronous USB interfaces are readily available and if your DAC doesn't have one, just grab an inexpensive converter like the CM6631(A) I used here. There are of course some very expensive USB-SPDIF adapters but given the CM6631A results, why spend more from a sonic perspective? As I have also stated before, I still have no good evidence to say that jitter at this low level is ever audible with real music despite what subjective audiophile reviewers or Internet posters generally claim! In ABX tests and even non-blinded flipping between measured jittery vs. clean audio sources, I cannot tell the difference at least up to ~2ns peak-to-peak jitter; nevermind the usual few hundred picosecond jitter with decent modern gear [Ed. apologize for the error...  I previously had a typo of "2ms"]. It's possible that my ears aren't sensitive to this but to be honest, unless one has superhuman hearing abilities, I can't imagine how it would be physiologically possible given these test results.

4. If you keep electrical noise/interference at bay, and use appropriate software player & drivers to ensure bit-perfection, there is no reason why Macs vs. Windows vs. ANY computer would 'sound' any different paired to a good DAC. As you can see above, when the CM6631A asynchronous USB-to-SPDIF converter is used and galvanic isolation is in place, there's essentially no difference in the analogue RightMark measurements and the Dunn J-Test spectra irrespective of CPU, speed, memory, HD/SSD, OS, USB, etc. (obviously as long as the specs are good enough for bit-perfect output - this is the beauty of digital). (Here's another interesting article showing identical Mac vs. Windows output using JRiver and different methodology.)

I hope this article has been useful in demonstrating a few things about computer audio. Measurements allow an opportunity to verify claimed improvements and demonstrate technical progress in hardware performance - amazing that few audiophile magazines / web sites actually include independent objective measurements and reviews come out looking like infomercials and PR. Realize that none of what I've shown is all that mysterious or requires audio "voodoo" whatsoever - science is all you need; let the musicians provide the passion and art. The cables are all generic, reasonably high quality (ie. nothing looking like they're going to break!), and universally costs <$20 for standard 3-6' lengths. The AUNE X1 DAC can be bought off eBay for ~$200. The CM6631A USB to SPDIF costs $50. The analogue and jitter measurements are on par with much more expensive gear I've come across. No need to further obsess over details like jitter or bit-perfection because the objective data has set the mind at ease...  If this were my main listening system, this then is where I would start more subjective evaluation with the amp and speakers asking myself whether I prefer the sound of the technically accurate setup or not. Realize that not everyone wants technical accuracy - terms like "clinical" or "sterile" impart this pejorative connotation. IMO, there is no such thing as a system being "too resolving", there's only source material that's "not good enough"!

After all this, you might ask: "So how does it sound?"
     Great!  (No need for superfluous adjectives.) Now go enjoy some music...


-------------------
For posterity, a response I put up on Audio Asylum questioning the results (April 21, 2013)...

http://www.audioasylum.com/forums/pcaudio/messages/12/122768.html

Gordon,
I appreciate your feedback and critique. I used to have the QB9 and indeed it is a nice DAC so was quite happy that Stereophile's measurements concurs.
For this article, I believe the data was "bit perfect" since the LSB jitter modulation was evident in the 16-bit tests. Due to the hardware limitations, this was of course not seen in the 24-bit graphs. I did discuss this fact and the issue that the built-in computer DAC's had problems with bit-perfection.

Thank you for bringing up the "skirt" issue. I'm going to have to update my ASUS Essence One measurements because this was worse with the USB input but not the coaxial or TosLink. However, most of what I've seen suggests that this low frequency jitter is masked by the primary signal. Please correct me if this is wrong so that I may pay more attention to it especially if there's a point of audibility.

As for "at some point you have to listen to the damn thing" (I'm directing this to the general viewership, not just to your comments). It is precisely because I've been listening AND reading forums like this for the last 15-20 years that I started doing the measurements and putting up the blog! I don't think it's at all a stretch to say that people are notoriously unreliable in their experience... I have never found an audiophile who could reliably differentiate high bitrate MP3 for example, yet when I ask, everyone seems to say they can (sure there are people who can ABX 320kbps but that's very few). For those who have a copy, listen to track 26 on Stereophile Test CD2 simulating 10ns jitter. I would estimate that half of the audiophiles I've tried this on over 50 years old have difficulty hearing this simulation on a test tone yet they universally claim that a few hundred picosecond jitter is a "problem" audible in actual music.

There's precious little objectivity out there anymore for audiophiles - look at that bizarre 3-part article on computer audio in TAS early 2012; if "everything's possible" (bit-perfect rips sound different if ripped at different CD speeds according to that article) then nothing is known anymore. Just look at the numerous comments when those new to computer audio asks straight forward questions here. IMO, some of the responses border on delusional.

Time is limited but I have spent many many hours listening to cheap to very expensive gear - even owning a few of the more expensive items over the years. It is my belief that "good" sound can be independent of factors like price, or any of the external factors like workmanship, etc. In fact, I've come across situations where high priced gear are just plain inaccurate. For me, accuracy is all that matters, and this can be quantified objectively.

Speaking of "misinformation is what is sending most people into shock when they start CA". Realize that over these months of testing, I have not advocated anything I would consider as "fringe". I've shown that asynchronous DAC's are better with less anomalous J-Test spectra, coaxial and good USB is better than TosLink in the gear I have, shown 'good' DAC's like the Oppo BDP-105 (ESS Sabre) measures superbly. I advocate for "bit perfect" and showed that both Windows and Mac are capable of accurate output, and have interspersed these tests with listening to make sure nothing appears awry - usually spending a couple hours at a time in the evenings listening (eg. when I was doing some cable measurements recently). As I've said in some posts, there is no voodoo or magic here, and the tests are showing me such. How in the world is this shocking anyone out of computer audio!? If anything, it's reassuring that measurements line up with a rational empirical approach that is reproducible.

Now as for the test gear I use and software like RightMark. You don't need the Hubble telescope to identify Jupiter in the sky. Likewise, why do I need expensive gear like an AP when I just want to make sure the frequency response is relatively flat, or that the dynamic range of a DAC can exceed 16-bit CD quality, or that the Dunn test isn't atrocious? All these things are within reach of what I have and the beauty of computer audio and technological advancement is that stuff that can do this is easily within the consumer's grasp. However, I'm not saying that anyone should do this since it has taken me countless hours to learn how to get the calibration right and make sure the software setup works for me in order to achieve a high level of reliability in the measurements.

When my simple setup can easily demonstrate analogue cable differences, XLR vs. RCA differences, show me the spectral smearing between 320kbps MP3 vs. 400kbps AAC, demonstrate J-Test differences between coaxial vs. TosLink vs. AES/EBU, when I post about this, in what way is this "misinformation"? In fact, my ASUS Essence One measurements showed a disturbing anomaly when upsampling is used resulting in attenuated frequency response which has since been confirmed by ASUS (and hopefully to be remedied)... How come subjective Essence One reviewers missed this when it was so obvious with just a little testing?

Look guys, ultimately either what I say and write about makes sense, or it doesn't. I mention a new post up here once awhile that could be of interest like this one with the laptop tests to outboard DAC's. I have ZERO financial interests or otherwise. I do it for fun, for my own education, and enjoy sharing what I've found with others. Anyone can freely share their thoughts in the comments section (haven't had to censor anyone at this point for nonsense), and I've invited people to give me evidence/reason if they think something I say is wrong. Of course, with my objectivist mindset, "Level 4 or 5" evidence (ie. individual or series of reports) doesn't really impress compared to experimental results or controlled trials.

Cheers,
Arch

MEASUREMENTS: USB Cables for Audio DAC's.

April 15, 2013, 10:33 pm

≫ Next: GUEST REVIEW & MEASUREMENTS: The Dr. Frank's "Best-Coaxial-Digital"™ SPDIF Cable.

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Up to this point, as you've seen in my measurements, I have been using standard generic shielded cables for analogue and digital. As you likely also know - if you've been involved in "high end" audio for awhile - there are few issues as contentious as whether expensive cables are necessary or if they even make a difference!

I've generally steered clear of this "debate" over the years because inevitably, these discussions with friends or forum threads on-line almost never really "go" anywhere...  Nobody leaves the discussion having really learned anything, and ultimately if the discussion turns into an argument, everyone just ends up feeling "pissed off".

The conclusion one usually can get agreement on is that the best cable is no cable. Cables are passive "components", and ideally should transmit signal without loss, and does not impart any kind of signature on the quality of the signal. This certainly makes sense especially in the analogue domain where noise can easily creep in, causing various distortions or raising the noise floor. But what of digital cables? When something as ubiquitous as a generic USB cable can easily transmit >20MB/sec data to a USB hard drive, how problematic can it be to get 150KB/sec across for 16/44, ~500KB/sec for 24/96 audio, or 'just' ~1.12MB/sec for 24/192 high-resolution audio!?

After all these months of testing and posting on various topics, I figure it's at least reasonable to explore the cable issue as best I can with (admittedly) the limited selection of choices I have available around here.

For the record, up to this point in time, the most expensive USB cable I have tried has been the AudioQuest Carbon (~$160 for 5') more than a year ago connected between the PC and a Benchmark DAC1 USB. I have since sold off both items. I could not clearly hear a difference with the AQ Carbon, but then again since the DAC1 uses ASRC, I really wasn't expecting much either. Since then, I've been satisfied with the selection of generic cables I've collected over the years as computer parts or 'freebies'. It is in this light that I selected among my assortment, 3 A/B USB 2.0 cables to test out:

USB Cable A:
- 3 feet long
- not gold plated
- ferrite core on the B end

USB Cable B:
- 5 feet long
- gold plated A & B ends
- no ferrite core

USB Cable C:
- 17-feet long! (6 feet Logitech extender, 5 feet Linksys extender, 6 feet generic cable)
- not gold plated
- possible ferrite core on the extender connections?

As you can see, Cable C was made by Dr. Frankenstein. It's unreasonably long (USB specs indicate a reasonable limit at ~16'), has 2 detachable connections in between, is not even gold plated, and who knows what kind of shielding is in that thing. Furthermore, the Logitech extender portion has the thinnest wire of all the cables. Any card-carrying "audiophile" would be laughed out of town for even suggesting they use something like this!

I. Cables tested with Asynchronous USB 2.0 CM6631A interface:

Setup:

Aspire Win8 laptop --> *test USB cable* --> CM6631A USB to SPDIF --> Generic TosLink --> AUNE X1 DAC --> silver braided RCA --> E-MU 0404USB --> Aspire Win8 laptop

Let us start with what has been previously demonstrated to be a very robust asynchronous USB interface; the CM6631A. Remember that the previous tests showed that despite different laptops used, the analogue and jitter spectra were almost identical.

With the different USB cables, RightMark DAC analogue analysis (24/96 used):

Frequency Response:

THD:

Stereo Crosstalk:

No difference.

Dunn Jitter Tests (16-bit & 24-bit tests):

Cable A:

Cable B:

Cable C:

No difference.

II. Cables tested with Adaptive Isochronous USB 1 AUNE X1 interface:

As shown previously, this interface can get noisy if I play and record with the Aspire laptop (see the laptop post). Therefore I'm going to use my ASUS Taichi to play the test audio while measuring with the Aspire as usual.

Setup:

ASUS Taichi --> *test USB cable* --> AUNE X1 USB1 --> silver braided RCA --> E-MU 0404USB --> Aspire Win8 laptop

RightMark DAC analogue analysis (16/44 only):

Frequency response:

THD:

No difference.

Dunn Jitter test (16-bit only):

Cable A:

Cable B:

Cable C:

Well, the adaptive USB interface is noisier and more jittery than the asynchronous setup as we have seen before in previous posts. Depending on which cable, we can see noise showing up at certain frequencies. For example, there's a -120dB 11.7kHz spike on Cable A, 11.3kHz for Cable B, 10kHz for Cable C. But these are not the symmetrical sidebands associated with data jitter. In fact, the low-frequency jitter associated with widening of the base around the 11kHz signal looks essentially identical and perhaps most importantly, there's really nothing unusual here to suggest that Cable C is of unusual length and construction!

Conclusion:

No evidence in these tests to suggest that the different USB cables used here in both the asynchronous USB 2.0 or adaptive isochronous USB 1 setups should sound different (even though one would expect Cable C to be the worst). Subjectively, listening to music with Cable C through Sennheiser HD800's sounded fine. 

No evidence with the J-Test to suggest data-correlated jitter is significantly different between cables. By the way, for a good demonstration of how jitter improves with interface/cable change, look at my post on Transporter-to-Behringer connection with TosLink vs. AES/EBU. Also, remember that my Oppo BDP-105 tests were done with a single 15' USB cable - still better than Cable C in construction :-).

As usual, feel free to drop me a note if there's good data or controlled tests to suggest USB cables make a significant difference contrary to these findings.

Happy listening...

GUEST REVIEW & MEASUREMENTS: The Dr. Frank's "Best-Coaxial-Digital"™ SPDIF Cable.

April 19, 2013, 1:45 pm

≫ Next: MEASUREMENTS: TosLink digital optical audio cables.

≪ Previous: MEASUREMENTS: USB Cables for Audio DAC's.

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By Keaton I. Goulden-Eyre III, Esq.

Two months ago, I had the good fortune to meet my independently wealthy friend Stephen at the local smoking lounge where 'the men' gathered to speak of recent developments in politics, finances, and what ails the common man. He mentioned that on a business trip to Yangon last year, he took a day off and was warmly received by representatives of Dr. Frankenstein's Audio Immersion Labs (FAIL) for a tour of a new manufacturing facility where only the best audio cables were to be produced by specially trained artisans by hand for export to rich Western nations where citizens truly appreciate the finer things in life. Within months, they were to start production of what was no doubt the best digital coaxial audio cable money can buy, to be followed by a full line of power cords, USB cables, and mineral-based sonic enhancers.

I was immediately intrigued! Post-haste, I contacted the legendary audiophile hardware importer Mr. Suet Shappeé in Luxembourg, the closest FAIL distributor to inquire about the state of the manufacturing facility and whether samples of their digital cable (code named "Siren") was available to audition.

"It's taking much longer than anticipated!" was his frank response. Apparently, the exact specifications for this product had just been cleared by the R&D-heavy company, run by the famous Japanese audio designer Dr. Seiji Minoeall working from his secret laboratory near Akihabara in Tokyo. Rumors are that the complexity of the computer simulations required to achieve the optimal design had to be debugged by the top 3 quantum mechanics physicists in the world working straight for 9 months!

"However, I am pleased to tell you the 3rd cable just came off the assembly line yesterday!" was finally Mr. Shappeé's response with much prodding on my part. Finally, after two hours of heavy negotiations, he agreed to send me a sample for review. Word on the street is that these cables are in so much demand that at the introductory price of $4999.99, it might take 1 year to satisfy the pre-orders! I felt honoured to be selected by FAIL as the first reviewer in the audiophile press to have access to this most rare and remarkable high-end cable.

Under strictest secrecy, I was directed to contact Minoeall-san for some further information about the product as well as design philosophy. "You know Keaton, audio is life. And like life, only the most complex can integrate the self-evident." I knew immediately I was speaking with one of the most learned designers I have ever conversed with. When he was young, Minoeall-san was raised by the monks of Mount Fuji where the simplicity of life taught him that the world around him was much more complex. On his frequent visits to the Kabukicho district of Shinjuku, Tokyo, he learned that in order to find truth and reality, one had to embrace the complex. It is with this wisdom that his creation was inspired. As fortune would have it, in 2001 Minoeall-san was visiting a high-end audio show when he met with representatives of Dr. Frankenstein Audio Immersion Labs and a joyous union was formed! "I have poured my life into the perfect design of this cable"; Minoeall-san confided in me during that intimate conversation.

Within 2 weeks of our discussion, an insured 50.8 pound Myanmar oak chest arrived at my chateau. The courier postman asked me - "what the **** is this? Gold?" Indeed, sir, pure audio gold!

I opened up the immaculately silk wrapped package to reveal the amazing new "Best-Coaxial-Digital" (aka Siren) cable - feast your eyes on the ultimate workmanship:

I will let Minoeall-san describe the exclusively patented 3-segment construction:
"Keaton, remember that in digital audio bits are NOT bits. There's just so much more that we've discovered in the last 20 years! Through deep metacognitive-quantum-mathematical research contracted with Top universities and the aerospace-defense sectors, we have determined that 3 is the ideal number - it is God's number - like the Holy Trinity.

"As more heads than one improves intellect, so too do extra connectors add to the coherence of the sound. This is why we used a triple-headed length on one end - this baby is totally overbuilt and even has the capability to conduct composite video if necessarily (note that the red connectors were used because red connectors were best to convey passion). We determined 10' as the optimal length for the first segment because it is. We then channeled the sound into a second triple headed 6' length where the audio is further refined and the well-recognized jitter is attenuated. We do this by making sure there is no gold plating since the gold-aluminum interface slows down the electron transition and induces timing artifacts. The final segment is a double-headed slim-line linear core design. It's only 3 feet long to speed up the final stretch for the electrons. The electrons will arrive at the DAC faster and this improves timing and coherence of the hi-fi signal, thus absolutely removing the jitter. With this new design, archaic concepts like impedance matching has become irrelevant.

"Some audiophiles may be surprised by the 19' length of this most wonderful of cables. As we all know, a short digital cable is bad. Too many electrical reflections distorting the leading edges of data transitions, confusing the DAC and creating digital errors. That is why the 'Super Connector' between each segment consists only of the best high-tech pliable organic polymer and aluminum. Any reflection will be subdued going across the connectors and rendered impotent.

"One more thing - gold is over-rated. It sounds bad because it's too shiny. It sounds digital - metallic and shiny like the CD. This is also why we do not believe in excessive shielding...  Remember, only complexity can convey the simplicity of truth - electromagnetic fields around us are complex, let us embrace this to finally create a cable worthy of the Best Music!"

Clearly, I realized this man spoke truth - finally, a sane voice in a cacophony of technobabble!

Subjective Analysis:

After taking that picture above and the unboxing video, it was time to plug the cable in. I connected it to my heavily modified Playstation 1 with digital output to another heavily modded tubed-NOS-DAC custom made for me by a local artisan-engineer consisting of specially selected chips, capacitors, and resistors based on auditory properties most engineers cannot comprehend. Likewise, my horn speakers and tube SET amplifier were custom made 'en bloc' by a one-armed carpenter in the city of Kuala Lumpur who specializes in the lost art of "tuning by ear". All analogue cables were of the rare cryogenically treated ninety-nine nines copper insulated by "active shielded" Super-Kapton connected as bare wires to the equipment with lengths conforming to strict golden mean ratios. Cable risers are of course essential. Obviously, only custom matching can achieve the synergy required for the finest audio reproduction.

I made sure to run the gear with the cable attached continuously for 2.5 weeks, getting my daughters to change CD's every couple of hours. This was only 420 hours and I am sure the "Best-Coaxial-Digital" Cable would continue to improve into the manufacturer suggested prerequisite 500 hour burn-in period.

To start, I put in the fantastic blues sound of Muddy Waters' "My Home Is In The Delta" (Folk Singer, Classic Records). In a word: stunning! My jaw and dentures almost dislocated. I have never heard so much air around the guitar in this 1964 recording. It was like being there marveling at the artistry of Mr. Waters. The resolution was so clear that I could envision the finger movements over each guitar string and the beads of sweat running down his cheek as he pined for that most beautiful of abodes - his home.

Next, I sampled the lovely rendition of "Love Of My Life" (A Night At The Opera, 1975, DCC) by Queen. By far, the 70's were the best! I met my 3rd wife when this song was released - great memories! Again, this cable did not let me down. Freddie Mercury has never been so romantic. Forget veils, my friends; never has an obstruction been parted like this since Moses parted the Red Sea. The liberation was epic. For a few minutes, I was 35 again with my girlfriend in my fifth generation turbocharged Chevy Impala.

For a taste of live music, I turned to Eric Clapton's "San Francisco Bay Blues" off his Unplugged (1992) album. What organicity! Not only could I hear every toe tap, wrong note, timing error, but I could literally smell the hint of cannabis wafting from the hooligans sitting 4 rows back from the stage. The soundstage enveloped my cubical listening room! I was astounded that so much detail could be preserved, retrievable only thanks to these digital cables.

Finally, I just want to speak of my favorite genre - classical music - the most highly refined of all genres. Obviously, classical music can only be appreciated with the best digital gear (but any analogue gear will do) due to the supreme intricacy and intimacy only this form of music can convey to the soul. Consider for example how totally inadequate any MP3 version of classical music becomes. I digress. I put on my favorite Mahler symphony - Symphony No. 9 conducted by Rafael Kubelik & The Bavarian Radio Symphony Orchestra (recorded live in Tokyo, 1975, re-released in 2000). The fine rendition of strings in the final movement contrasted lovingly with the percussion in the Rondo-Burleske so perfectly that my eyes welled up in tears, gave me palpitations, and triggered my pacemaker. How sad that Mahler himself never lived to hear his final angelic masterpiece. At one point, my luscious young wife Candy poked her head out of the drawing room and commented "that's a fantastic vinyl!" - I grinned like the Cheshire Cat.

To confirm these findings, I invited my best friend, the musician Jonathan C. Wiltkinshireshire II to hear these cables. As we listened over a glass of fine single-malt scotch whisky and puffs of freshly imported Cohibas, he commented on the "resolute ease" of presentation and "perfect liquid presence" of these cables. I could not agree more.

Of course for the remaining precious moments with the "Best-Coaxial-Digital" SPDIF Cable, I auditioned numerous landmark recordings from Chesky, Blue Coast (my only regret was that I could not audition in pure DSD), MoFi, Audio Fidelity, Classic Records, Reference Recordings, AIX, 2L, B&W, Linn, etc. My cup overfloweth with audiophile delight thanks to this cable. One parting observation as I begrudgingly conclude these "subjective" comments - I have never heard Rebecca Pidgeon's shakers ever go that "deep" into the soundstage on "Spanish Harlem"!

Objective Analysis:

First, let me make it clear that this section is utterly unnecessary. Everyone knows that my ears are superior to all other imperfect measurement devices (after 73 years of life experience, I have heard every relevant song ever produced by any audiophile label and am certain that my auditory acuity and frequency response have never been better). It is because my Commoner-associate Archimago demands that this section be included that I am even wasting my time documenting this. It's insane that anyone can come to any conclusions using free software and decidedly "pro" audio hardware! Nonetheless, let us obtain some numbers...

Firstly, let me introduce you to the inferior comparison "digital" cables:

Cable A is a feeble short 3' length of shielded coaxial from Radio Shack Canada before they went defunct a number of years ago. Cable B is a mass market 6' "digital audio" cable from Acoustic Research produced in China. I bought this one 13 years ago so I am confident it's worthless since the science of digital audio has moved ahead multiple generations. Decidedly pedestrian.

Cable C is insanely long at 25' marketed by Acoustic Research as a shielded "subwoofer cable". Obviously this is an analogue cable and I would dissuade anyone from considering this in the digital domain. I noted that there is "directionality" with some arrows pointing the flow of electrons. Despite my protestations, Archimago insisted I hook the cable up backwards (the man is truly an anarchist).

So, using his audiophile unapproved "usual" testing hardware:

Win8 laptop --> 6' shielded USB cable --> CM6631A asynchronous USB-to-SPDIF --> *test digital cable* --> AUNE X1 DAC coaxial input --> silver braided RCA --> E-MU 0404USB --> Win8 laptop

RightMark Audio Analyzer 6.2.5 Summary (24/96):

Preposterous that the beauty of musical hardware can be reduced to "bels". Seriously, bells are so passé (especially the tubular variant), they barely play a significant part in any respectable percussion section.

Frequency Response:

Noise Level:

THD:

 

Stereo Crosstalk:

That purplish color for the "Best-Coaxial-Digital"™ is clearly regal and hence the best. Cable A was too short, Cable B was too cheap, and how dare anyone hook Cable C up against the clearly marked electron directional flow and expect good results?!

Jitter analysis (16-bit and 24-bit):

Cable A:

Cable B:

Cable C:

"Best-Coaxial-Digital"™ SPDIF Cable:

Squiggles, squiggles everywhere. Who know what this means, but I am sure the "Best-Coaxial-Digital"™ performed superbly compared to the other questionably designed plebian pocket-change cables in the jitter.

Conclusion:

As you can see, the objective results were utterly pointless and do not explain the sunny vs. total eclipse audible difference whatsoever. Anyone who listens without noticing a difference must have sensorineural hearing loss. As I told Archimago, the complexities and incalculable dimensions of music are so expansive, one cannot possibly expect a few simple measurements, ABX, DBT, ABC, NBC, CBS or CNN efforts to have any correlation with hardware quality. I echo those wise words of Minoeall-san: "only complexity can convey the simplicity of truth". My wife knows this, my audiophile friends know this, the righteous gentlemen of the audiophile press are well aware of this truth and have done their utmost to educate the unsophisticated masses. Even my 4 year old grandson told me the other day "hearing is believing, grandpa." Yet I digress again...

This fantastic coaxial SPDIF cable has earned the KASH (Keaton's Audiophile Superior Hardware) Jadeite Studded Platinum Award and rightly deserves to be included in any A++++ list of digital gear. In fact, it's so good, I wish I could create a category called "semi-analogue"! It was with great regret that I had to pack up these cables after this review to return to the manufacturer. Due to élite availability, this set I borrowed had to be sent off immediately to a billionaire buyer in Beijing who was glad to know that I had almost completely burned-in these amazing cables. Very soon, Mr. Suet Shappeé in Luxembourg will be opening a showroom (Delüsion Audio Arts GmbH) to highlight the full line of Dr. Frankenstein's Audio Immersion Labs gear and any serious audiophile would be well advised to experience this cable for themselves. You will be utterly enraptured!

One last detail, the introductory price of $4999.99 will only be available until December 31, 2013 (increasing to $5888.96 in 2014). For the level of performance, this is a fantastic deal. Minoeall-san was firm regarding the pricing when he told me "I cannot sell this for a dollar less if I want to recoup my R&D costs!" I am astounded by the honesty and philanthropy demonstrated by this company.

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Ed: Enjoy the music.

MEASUREMENTS: TosLink digital optical audio cables.

May 1, 2013, 7:39 am

≫ Next: MEASUREMENTS: Analogue RCA Interconnects.

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Let us now finish off testing the digital audio cables at my disposal. Already we're seen that USB and coaxial SPDIF cables did not have any measurable differences even with poor construction and length. Remember that the benefits of TosLink ("Toshiba Link") is that they are non-electrical so have the benefits of being resistant to electrical interference. Furthermore, it has been said that they can be used in long runs without loss (specifications allow up to 33 feet).

I have a few old TosLink cables hanging around of varying qualities and build. Let's have a peek in the closet:

Cable A:
This is the cheapest cable I have. I believe it was a freebie from one of my old DVD players. The only marking is along the side of the cable it says "VITOnet". Length 6' and relatively thin.

Cable B:
A small step up from the freebie. Branded "Thunder Cable" TosLink I got as a package with some other home theater cables I bought maybe 10 years ago from Costco locally. Again 6' and thickness of the cable about the same as the VITOnet, so it feels quite fragile in the hand.

Cable C:
Another step up in construction and price. A 6' Acoustic Research Pro TosLink cable. Nice gold plated connector now with good strain relief and quite a thick caliber. Maybe about $20 when I bought it. Has served me well for years between the DVD player and a Denon receiver.

Cable D:

Up to this point, all the cables have been plastic optical fiber. This next one is of glass cable construction. Can be purchased off Amazon here for ~$20. Construction is also very good with nice metal connectors, gold plated terminals, and good cable thickness. According to the "specs", the cable supposedly has "280 individual stands of glass". Again, the length is the same at 6'. Note that it's not necessarily a given that glass is superior to plastic; for LED light sources, it has been said that plastic cables might even be better.

Cable E:

The "worst" Dr. Frankenstein cable I could construct from what I had was with the use of a $3 TosLink coupler off eBay. This adds an extra transmission interface and of course extends the length of the cable which could worsen signal integrity. I coupled the 2 cheapest TosLink cables I described above - the VITOnet (Cable A) and Thunder Cable (Cable B) - for a total length of 12'. I tested the cable looking just like this with it coiled up.

Analogue Tests (RightMark, 24/96):

Standard methodology:

Win8 laptop --> 6' shielded USB cable --> CM6631A asynchronous USB-to-SPDIF --> *test TosLink cable* --> AUNE X1 DAC TosLink input --> shielded RCA --> E-MU 0404USB --> Win8 laptop

Okay folks, here's the summary...

I think as you can see from this table, the test results are essentially exactly the same. 'Cable E' was measured about 2 weeks after all the others (I was waiting for the coupler to arrive from Asia); notice how reproducible the test results are.

Frequency Response:

Noise Level:

THD:

Stereo Crosstalk:

No difference.

Jitter Analysis (Dunn J-Test Signal 16-bit & 24-bit):

Cable A (6' VITOnet):

Cable B (6' Thunder Cable):

Cable C (6' Acoustic Research):

Cable D (6' Glass):

Cable E (12' Coupled VITOnet + Thunder Cable):

For the 12' Cable E, I even started waving the cable around at the coupler joint to see what would happen to the jitter spectrum... Nothing of significance!

Notice that the CM6631A USB-to-SPDIF converter has quite low jitter overall despite this being TosLink - few nasty sidebands and little spectral spreading of the primary signal (usually in my other measurements TosLink not as good as the coaxial interface).

Conclusion:

TosLink cables measure the same based on these results. Inexpensive 6' plastic, better quality plastic, glass, or a longer stretch of 2x6' cable with a coupler did not show any measurable difference in the analogue output from the AUNE X1 DAC used to test. Likewise, jitter analysis with the Dunn J-Test in both 16 and 24-bit versions did not demonstrate any appreciable difference between the cables. As far as I can tell, jitter does NOT appear to be affected by the TosLink cable quality based on these tests.

Subjective listening to the 12' coupled Cable E likewise did not show any musical anomaly - was able to enjoy Jorma Kaukonen's Blue Country Heart thoroughly through the AKG Q701's :-).

I think this concludes my survey of digital cables - at least for now...

MEASUREMENTS: Analogue RCA Interconnects.

May 2, 2013, 7:42 am

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≪ Previous: MEASUREMENTS: TosLink digital optical audio cables.

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Now that I've measured the main digital cables used for audiophile listening these days (USB, coaxial, TosLink), I figure it's time to demonstrate what happens when we bring the testing methodology with analogue cables. [I'm hoping in time HDMI will make some headway into the audiophile DAC world since I look forward to multichannel playback one day through my computer server.]

Remember why we got digital in the first place: robust data storage free from transmission and generational losses - in other words, resistance from corruption.  By transforming data into 1's and 0's, we quantize the data into binary form and complexity is thus encoded in larger quantities and combinations of this quantized binary data which can be saved in a form which makes detection and correction of error possible.

As a result, when digital 'works', it likewise tends to be 'all or nothing'. What we saw with the digital cables is an example of this. Despite some really poor quality cables tested (the Dr. Frankenstein models!), the measurements were essentially identical in each case (they all worked). Parameters like impedance and capacitance of the cable do not affect the transfer of data unless of course they are outside of tolerance for the interface. It's of course possible that the occasional 'bit' of error occurred with poor cables, but obviously it was not large enough to encroach on the measurement results (nor affect the subjective audibility when I was listening). Beyond "bits are bits", in the timing domain, there have been reports of very long cable lengths potentially worsening timing of the digital signal (ie. jitter), what I have seen suggest lengths of 20-30m AES/EBU adding maybe 20ps - totally irrelevant to audio performance and at lengths we generally do not use in the consumer setting.

Let's see now how good ol' analogue interconnects of various lengths fare.

Here are the models being tested:

Cable A:
3' freebie RCA cable that came with an old cheap DVD that has since broken. Connectors not gold plated.

Cable B:
3' Radio Shack shielded RCA cable. Gold plated connectors.

Cable C:
6' Radio Shack shielded RCA cable. Gold plated connectors.

Cable D:
10' poorly shielded composite cable (with stereo audio). Gold plated connectors.

Cable E:
3' pure 4N silver, 4-core braided interconnect. Connectors are gold plated Neutriks. Soldered with Cardas Quad eutectic silver solder. Got this cable about 2 years back and used for my SACD player from here.

Cable F:
With apologies to Keaton & Minoeall-san, I used the 'Super Connectors' from the "Best-Coaxial-Digital" cable along with the two lengths of 10' composite cable and 6' stereo RCA and created an un-audiophile-approved 16 foot RCA "double cable" interconnect.

Setup:
Win8 laptop --> shielded USB --> CM6631A async USB to SPDIF --> 3' coaxial --> AUNE X1 DAC --> *test analogue interconnect* --> E-MU 0404USB --> shielded USB --> Win8 laptop

Summary RightMark 6.2.5 results (24/96):

Frequency Response:

Noise Level:

THD:

Stereo Crosstalk:

Summary:
1. Analogue ain't digital! Although in most ways the measurements are very similar (these are short lengths of interconnects after all), mild differences can be found.

2. Frequency response unchanged among the cables. Interesting. Some people talk about analogue cables as "tone control". I don't see it using these interconnects even with longer length (there is a hint of high frequency roll-off with the 16' cable but really this is trivial) or different conductor material. Using silver interconnects, there are no changes in the frequency response to suggest these cables sound "brighter" as some contend :-).

3. Interesting Stereo Crosstalk performance. Stereo crosstalk looks to be sensitive to cable length. The silver cable had the least crosstalk up to 5kHz and then increased from there - this is possibly a function of the fact that it's constructed as 2 separate cables as pictured above rather than the zip-cord arrangement of the other cables.

4. Measures like THD should not (and in fact does not) show a difference. After all, cables are passive "components" so should not introduce harmonics into the equation. As for noise floor, I suspect if I were to test under conditions with strong RF noise the poorly shielded cables would perform worse (may try this later), but in the home environment where I tested, obviously this was not a problem even in reasonably close proximity to the laptop, DAC, and E-MU ADC.

There you go. Analogue interconnects do make a slight difference and this is quite measurable particularly in terms of stereo crosstalk performance. Remember that these interconnects are of relatively short lengths so minor differences are really not surprising. The obvious question is - would humans be able to differentiate these interconnect cables based on listening tests? I honestly doubt it. Subjective listening using my test setup did not reveal any noticeable change with the long cable vs. the short silver cable. Realize that even with the long 16' cable, stereo crosstalk was still below -75dB which should be inaudible - for comparison, high-end LP cartridges are only capable of 30-40dB crosstalk performance.

Musical selection tonight was Rachel Podger & Brecon Baroque's renditions ofBach's Violin Concertos on Channel Classics (SACD converted to 24/88). Sounds great with the 16' Frankenstein cable with my Sennheiser HD800 headphones off the E-MU 0404USB.

MUSINGS: "Audiophile" Digital Cables...

May 3, 2013, 3:12 pm

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≪ Previous: MEASUREMENTS: Analogue RCA Interconnects.

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As I noted previously, the purpose of going digital has to do with prevention of signal corruption. In doing so,  we can speak of "bit-perfect" transmission of audio data in a way which is impossible in the analogue realm. For example, if we think about the main "container" of consumer high quality analogue music today - the LP / vinyl - nothing can be considered 'perfect'. Each LP is slightly different in terms of being free from warping or (hopefully) minor groove imperfections from the moment it leaves the pressing plant (even the quality of each stamper used varies by generation and age), each time the LP is played, a little bit of damage happens to it so every playback will be different. Even if you keep it in pristine condition, the ravages of time and environmental factors will take its toll on the material itself in large or small ways. Furthermore, there is no way to replicate the music in 'perfect' form as a result (unless you digitize it of course; but only to the quality of the LP playback gear and ADC).

As we know, the CD technology (~30 years old now!) is different. Small imperfections in the plastic or aluminum 'pits' do not lead to audible anomalies thanks to the Reed-Solomon error correction of the digital data. Furthermore "bit-perfect" ripping is routinely done (obviously to the horror of the music industry over the last few decades) and the result is literally innumerably perfect copies.

Let us turn our attention to "digital cables". For some reason, some people seem to forget the above and believe that different digital cables make a difference to sound even when there is no disagreement that bit-perfect data is being transmitted down the pipe in digital form. In fact, there has never been a plausible explanation provided by the supporters of different digital cables. Some talk of jitter being inherently different between cables, some that perhaps electrical noise will disrupt the phase transitions in a digital signal to worsen this jitter and perhaps degrade the bits. But where is the evidence for such beliefs beyond subjective "impressions"?

An industry has developed around digital cables of all sorts. Some of these cables are extremely expensive with price tags of hundreds to thousands. Purported benefits include more exotic / precious conductors, various types of insulation, winding techniques, upgraded connectors. While this is well and good - nobody denies that a well built USB cable with excellent strain relief and connectors that do not break or oxidize is undesirable - why is it that so often audio quality gets thrown into this mix of features during discussions as if it's some kind of 'given'? If a friend says he/she bought a $150 cable that was well shielded, had great connectors, was the right length, and had great cosmetics, I do not believe I would question the motivations since these are all quite reasonable. But if they said "this $150 cable sounds better than all my less expensive ones", I think it would be reasonable to at least wonder about the truth of such a statement just as much as if the friend said "water tastes better from my wine glass compared to just a regular glass cup". Has there EVER been a proven example of a properly functioning digital cable sounding different from another item of the same type?

When it comes to cabling in general, I find it interesting that talk of sound quality is usually propagated by the audiophile press and audio reviewers on-line rather than directly from the manufacturers (see the addendum below for an example of a cable ad). False advertising is a legal offense after all while reviewers spouting off their opinion is given artistic license as subjective experience. This is all fine I suppose so long as it's "above board", but in a small industry where ad revenue is of major importance in the print magazines and web sites, do we really think there is a significant firewall between the reviewers and the source of financial support? A reminder - cables are one of the highest marked up items for any manufacturer. I wonder just what percentage of a cable manufacturer's budget goes into R&D vs. advertising...

Turning to more "objective" matters, let us recap. Over the last few weeks, I measured a number of digital cables of various types in my standard test setup:

USB cables:
http://archimago.blogspot.ca/2013/04/measurements-usb-cables-for-dacs.html
S/PDIF Coaxial cables:
http://archimago.blogspot.ca/2013/04/guest-review-measurements-dr-franks.html
S/PDIF TosLink cables:
http://archimago.blogspot.ca/2013/05/measurements-toslink-optical-audio.html

In each case, using what I had, I tried to "create" improbable setups (eg. very long cables, use of couplers and extenders) which serious audiophiles would likely feel will deteriorate the signal quality. Yet in NONE of these situations was I able to detect an actual loss of fidelity running test signals which have been successful in detecting various types of anomalies over the course of my postings. Subjective listening likewise did not suggest to me any deterioration in sound quality. In contrast, RCA analogue interconnect measurements can be shown to have subtle differences even with relatively short lengths.

What else is there to say, really? In my opinion, there is no other way to interpret the data than to conclude that digital cables of adequate quality to transmit the data in a bit-perfect fashion makes no difference to sound quality. Furthermore, there is no evidence to suggest that the cables influence jitter to any significantly audible degree (again, "jitter" tends to be the scapegoat for almost all digital imperfections for selling hardware). Yes folks, bits are bits as far as the digital cable is concerned. They were engineered to be like this. Over the years I have tried my hand at listening tests with digital cables; but never once have I been able to convincingly differentiate cable quality in any blind testing.

Of course, I could be wrong and open to this possibility...  As usual, if you have good data to show differences between "working" digital cables, please leave a comment.

Enjoy the music...

Addendum:
Here's an interesting advertisement showing "proof" of something (I reproduce this as 'fair use' for the purpose of commentary with no expectation of benefit to myself financial or otherwise):

For the purpose of discussion around just the contents of this ad, I blurred out the company information and trademarked names...  You're welcome to check out the ad found on page 45 of the January 2012 issue of The Absolute Sound (same issue as one of the most disinformative series of articles about computer audio ever to 'grace' an audio magazine from "Dr. Charles Zeilig and Jay Clawson").

1. These cables meet or exceed $20K-$60K competitors' performance. What length are these $20K-$60K cables??? What cables cost this much? Are we talking individual cables or cabling a whole studio? All I can say is throwing a huge number out like that could be either "impressive" or absolutely ridiculous.

2. What is this "patent-pending" XXX Technology? Is it metallurgical? Connector +/- plating? Cable topology? Insulation material? Such a mystery! (To be fair, you can read more about it on the web page to some degree and ponder the claims - go see the ad.)

3. "Nice" plots of square waves I guess...  So it's "absolute proof" that XXX Technology improve rise and fall times of a square wave. We know nothing of the length, type of wire, or even what changed between the "conventional cable" compared to the "same cable" with XXX Technology. Yeah... Nice and "scientific" looking but totally lacking in transparency!

4. What does that square wave plot mean for audio, exactly? Are they trying to say these are digital cables where more precise square wave transitions might imply less timing inaccuracies? But we see analogue speaker spades and XLR connectors (I suppose could be AES/EBU).

5. Notice there's no claims of sound quality. Even the TAS and Rick Rubin quotes say nothing about sound quality, rather something nebulous like "performance". Interesting, given that TAS reviewers commonly speak of comparative sound quality in their reviews of cables yet the company chose not to include such quotes, why?

6. Notice the nice arrows showing directionality of cabling for the speaker spades...  How thoughtful.

Things that make you go... Hmmmmm...

MEASUREMENTS: Power Cables for Low Power Audio.

May 6, 2013, 12:03 pm

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We are told - "everything makes a difference!"

Expensive power cables are an example of taking this principle more than likely to the extreme - well into the territory of the neurotic obsessive-compulsive. Some audiophiles claim there are very significant differences to be found by replacing standard cables like the common IEC connector varieties between the mains and one's gear. DIY plans are available on the Internet, and of course many enterprising companies have produced all kinds of cables to satiate those "believers". Like other cable claims, it's difficult to determine what scientific / engineering theory could account for these beliefs. While there could be some justification to use of heavy duty power cables for high-powered amps with dedicated circuits for example (very rare for home audio), why would someone need fancy cables for devices like DAC's or CD players where internally the AC is converted to low voltage and current DC to power the electronics? Furthermore, we all know that the electricity supplying our gear is connected by hundreds of miles of plain old non-"audiophile approved" copper cables of various diameter and quality.

In order to look for tiny differences, I'm going to try using various power cords with the ASUS Essence One DAC (note that my DAC is slightly modded with all LM4562 op-amps)... Let's see if there are any differences looking at the analogue output and changes to the J-Test jitter spectrum.

First, as usual, I had a look into my closet of cables to see what I have. Here are today's selection:

Cable A:
No nonsense generic freebie 6' cable that came with my old Antec computer power supply. Has the brand name "LINETEK" stamped on the connector.

Cable B:
Notice the green dot on the plug. That means this is a higher quality "hospital grade" cable. Also 6', but it's about 25% thicker, and twice the weight of Cable A. Strain relief is fantastic. The metal wall plug prongs are more substantial and the ground prong is solid metal instead of hollow like for Cable A. Presumably the thicker diameter indicates better shielding. I know this particular brand of cable is being used in the local hospital's ICU department. If this cable fails during use, patients could die...

Cable C:
I looked around to see what was the absolute WORST power cable I could come up with. Here it is - total 56' long. Using Cable A, I connected it to a 50' yellow outdoors cable I used over the Christmas holidays for the outdoor lights. In fact, this cable has been used for this purpose for the last 5 Christmases at least, so it's been exposed to the dirt, rain and snow. The metal prongs in fact look worn and oxidized. In fact, this is so nasty that I took a picture of it out on the deck since my wife refused to have it indoors for more than a few minutes for testing :-). I tested it connected to the DAC pretty much looking like this tangled mess. Unless you think the last 6' of generic power cable can make a difference, the "performance" of this cable should unequivocally "sound"/perform terribly.

Gear Setup:
I used a variant of the usual testing setup:
Win 8 laptop --> shielded USB --> CM6631A asynchronous USB to SPDIF --> Acoustic Research 6' TosLink --> ASUS Essence One (*connected to wall outlet by test cable*) --> 6' XLR cables --> E-MU 0404USB --> shielded USB --> Win 8 laptop

Note that I decided to use the CM6631A device for USB input and TosLink out (previously tested) instead of the native Essence One USB because I actually found less jitter this way. I noticed that the Essence One's USB input has a fair amount of low level jitter artifacts - not sure if it's a result of the CM6631 (non-A) chipset or the drivers in this configuration.

Analogue Measurements (RightMark Audio Analyzer 6.2.5, 24/96):

Summary:

Pretty much identical...  Very small differences within the error range for each "run" of the test.

Frequency Response:

Noise floor:

THD:

Stereo Crosstalk:

As you can see, there's nothing here to differentiate the analogue measurements from the DAC using the different power cables.

Jitter Analysis (Dunn J-Test - 16-bit and 24-bit variants):

Cable A - 6' generic:

Cable B - 6' Hospital Grade:

Cable C - 56' - 50' outdoors corroded prongs + 6' Cable A:

Again - no real difference folks. Not really that one expects any difference since it's unlikely that the DAC's internal timing circuitry could be affected by the AC input. Note that with the Essence One, we can actually see the 24-big jitter modulation pattern due to the very low noise floor below -140dB.

Conclusion:

As usual, I listened to the audio output using the poorest cable configuration after I ran these tests and as I ponder what to write for the blog entry. Indeed, the sound was fine.

Tonight, I was listening to the Erich Kunzel & Cincinnati Pops' rendition of Tchaikovsky's 1812 Overture (Telarc 2001, SACD digitally ripped & converted to 24/88) with the Essence One powered by the nasty 56' length to the wall socket. It sounded good. By ~14 minutes into the track, we hear a multi-textured climax with church bells, choirs, brass, percussion and of course cannons. The complex mix was reproduced very well and rendered nicely with my AKG Q701 headphones - plenty of dynamics being pumped out into the Essence One's headphone amp.

Sure, it's possible that "everything makes a difference!" As in most things in life, the wise man needs to ponder the claims a little further to divine the truth. At least when it comes to power cables, I think the wise man can comfortably walk away from such claims of audible differences and realize that a decent IEC cable is all that's needed - at least for low power devices like a DAC.

As is my usual policy, I do not bother measuring high-priced cables - partly because I don't have any at home - but these posts are not about pointing fingers at specific companies. Rather I hope the measurements and comments stimulate thought. Note that I have "heard" expensive power cables over the years so am well aware of their "performance". As usual, drop me a note if you have good evidence to show otherwise...

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INITIAL IMPRESSIONS: TEAC UD-501 USB PCM & DSD DAC (Part 1)

May 9, 2013, 10:09 pm

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This guy arrived at my doorstep on May 7. Over the next week or so, I'll just build up this blog with TEAC UD-501 information as I gain experience with the unit.

Initial Impressions & The Basics:
By now, you would likely have seen the specifications sheet on this device if you've been researching.

It came relatively well packed in the box. I paid the current market price ~$850USD. Standard styrofoam protectors to withstand bumps and thick plastic bag around the unit itself. Inside the box is just a standard decent IEC power connector, an instruction pamphlet I didn't even look at and a really unimpressive thin zip-cord RCA cable :-).

The unit itself IMO looks great as do the line of "Reference Series" gear - utilitarian in terms of knob and display placement with a hint of the TEAC heritage with "pro" gear given the side metal handle bars - looks like rack-mount gear. Remember that TEAC [Tokyo Electro Acoustic Company] Audio is in the same family as Esoteric (consumer audiophile) and TASCAM (pro audio); depending on how you look at it, I guess it's either an upscale TASCAM without all the plastic or 'baby' Esoteric without as much of the mass and audiophile aesthetics.

The weight is quoted as 9lbs and it certainly feels substantial. It's about the size of an A4 (letter) sheet of paper (front "handles" poke a bit forward) and 3 inches high. The construction is metal all around with a nice brushed metal texture so there's no shiny bits - nice. Knobs feel very stable and responsive. The headphone knob on the right rotates smoothly and the MENU button feels authoritative when pressed (unlike the front buttons for the ASUS Essence One - just one of those subjective look-and-feel things which adds to a positive impression).

The organic electroluminescence (essentially OLED) display is easy to read, has 3 brightness settings and an "OFF" setting. I like the amber color which is non-distracting and I made sure I set the default to the dimmest setting. Great also that the amber LED for input selection isn't too bright and certainly less distracting than the Essence One's blue LEDs (not a big deal for me but I know many folks get bothered by this).

Other than to get more detailed descriptions of the menu options, the manual is quite unnecessary - it's really easy to operate... Basically push the MENU button to toggle between options, turn the left knob to change selections, that's really it. In looking over the menu selections, one cannot help but think that the TEAC engineers basically took the TI/Burr Brown PCM1795 DAC chip, looked at the datasheet - considered the undocumented modes, and created a machine that took advantage of everything this DAC chip can do! Here are the main options:

1. PCM Upconversion to 24/192 - presumably could help reduce jitter.

2. PCM1795 digital filters: SHARP, SLOW, and OFF - hadn't seen the OFF option before; an interesting mode which I believe was intended to allow the DAC chip to be mated to an external filter.

3. DSD Analogue FIR filters: FIR1 to FIR4 - I'll discuss more about this when I present the DSD measurements.

4. Analogue output: either RCA, XLR pin 2 hot, or pin 3 hot. Cannot output both RCA + XLR.

5. Simultaneous headphones + analogue line out: ON or OFF.

6. USB input power - powers off the USB port if another input being used - not sure the reason for this, actually, just power saving I guess?

7. Setting mode display: ON / OFF for the display to show if upconversion is happening, PCM / DSD, sampling rate... Very cool. I leave this ON.

8. LCD dimmer - 3 levels & OFF.

If you look at the PCM1795 datasheet, you see that it's documented to be a 32/192 part and can do DSD64 (2.8MHz) conversion. Perhaps a little known fact is that this DAC chip is capable of 32/384 PCM and DSD128 (5.6MHz) as "undocumented" features which the TEAC designers obviously capitalized on. Note that the ASUS Essence One also uses the PCM1795 and "symmetrically upsamples" to 24/352 or 24/384 depending on whether the input sampling rate is a multiple of 44kHz or 48kHz.

So far, the Windows driver 1.02 seems quite stable. No problems with ASIO PCM using foobar2000, and DoP bit-stream support through JRiver 18.0 works well for DSD. The current TEAC HR Player 1.0.0.4 (small basic music player, "portable" so no install) works to play back DSD and can stream using either DoP or "native" ASIO 2.1. If you have DST lossless compressed DSD audio, the TEAC player doesn't seem to handle these but they're fine with JRiver.

On the Mac side (MacBook Pro with Mountain Lion), it uses the standard USB Audio 2.0 driver so nothing to install. I have used both Decibel for PCM playback and the "alpha" JRiver 18.0 for Mac works essentially the same as the Windows version for DoP support.

Subjective Sound Quality:
So far most of my testing has been with the Sennheiser HD800 pictured above. I'm just going to put on my "subjective reviewer" hat for a moment...

The headphone amp sounds good. It's not powerful - rated at 100mW into 32 ohms but it drives the HD800 loud enough including some relatively soft classical test tracks I had. The amp could easily drive these headphones to ear-splitting levels with the usual commercial rock/pop/jazz/country tracks. The AKG Q701's are a bit more difficult to drive so I would avoid using these with softer classical selections with the TEAC. It's quite clear that the ASUS Essence One has a significantly more power headphone section in comparison. (Of course if you're a big time head-fi fan, TEAC would want to interest you in the HA-501 headphone amp.)

So far, I have no complaints of the sound. PCM performance is excellent. For example, a test track I often use to weed out poor systems is Tyler Bates' "To Victory" from the 300 (2007) soundtrack. It's recorded "hot" and dynamically compressed and the cacophony of sounds tends to get muddled very easily on a poor system. This track was reproduced excellently with this DAC (I also find the emotional response - that sense of dread - conveyed by this track a good personal gauge).

On the Kodo track "Niji No Nagori" off the Tsutsumi (2000) album, there's a nice build up of multi-layered drums, flute, vocals, culminating in a woman singing with clapping, percussion, and male backgrounds around 5:00. The drums sounded dynamic and "full". Bass went deep with the HD800; and thanks to the "speed" of these HD800's, it sounded precise. Again, excellent performance and I would certainly rate this DAC+headphone amp highly.

Currently, I don't have much DSD music collected yet but have ripped a number of my SACD's which I know are either DSD sourced or high-resolution analogue in origin - no PCM or worse Red Book-sourced DSD for me like in this review, thanks.

Albums heard or tracks sampled: old analogue sourced Nat King Cole's The Very Thought Of You (Analogue Productions 2010), Pink Floyd's The Dark Side Of The Moon (2003 remaster), Michael Jackson' Thriller (1999 remaster), Al Di Meola et al. Friday Night In San Francisco (1997 remaster), Miles Davis' Kind Of Blue (2007 Japanese SACD). They sound good overall...  Limitations of the analogue source quite evident with obvious limited noise floor on most of these. The 80's sound of Thriller is pretty dated but I think the SACD version is the best sounding 'pressing' I've come across...

Modern DSD sourced SACD's: Erich Kunzel's Tchaikovsky 1812 (Telarc 2001), John Hiatt's Master of Disaster (2005), Jorma Kaukonen's Blue Country Heart (2002), Rachel Podger's Bach Violin Concertos (2010), Stuttgarter Kammerorchester' Die Rohre (Tacet 2003). Nice, clean, great sense of space especially the Stuttgarter and Rachel Podger SACD's.

There's very little DSD128 content out there as far as I am aware...  However, I downloaded a few of the samples from 2L. They sound excellent but since they're sourced from DXD (24/352), I could also download those massive files (1GB for 10 minutes!) and play them PCM direct and be even "closer" to the performance :-). Seriously folks, I think this would be a real waste of disk space!

You may be asking - is there anything "special" about the sound of DSD - especially after I penned this piece on DSD? Well, honestly, it's hard to say... Really hard to do any kind of direct comparison since the foobar200 ABX tool doesn't work for this, and the switch from PCM to DSD results in a soft 'click' sound as well as a brief delay...  Furthermore, volume levels aren't exactly matched. All I can say is the music just sound good whichever format :-). I don't think DSD is "needed" for good sound, but it's nice to be able to play back the music in whatever the original format was without transcoding.

I'll be back this weekend with some PCM measurement results...

Links to the objective evaluations:

PCM Evaluation (Part 2)

DSD Performance (Part 3)

MEASUREMENTS: TEAC UD-501 PCM Performance (Part 2)

May 11, 2013, 8:28 am

≫ Next: MEASUREMENTS: TEAC UD-501 DSD Performance (Part 3)

≪ Previous: INITIAL IMPRESSIONS: TEAC UD-501 USB PCM & DSD DAC (Part 1)

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Okay folks, let us continue with the TEAC evaluation... First, we need to look at the PCM performance of this DAC. Although DSD may be the "hot" feature of DAC's these days, PCM remains the most important digital encoding method. A good DAC MUST perform well with PCM music.

General setup:

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

The TEAC has quite a "hot" XLR output and unfortunately clips the E-MU so I was unable to get an accurate reading without using volume attenuation (rest assured it does look very good, dynamic range is probably about another 6dB better that what I got with the RCA 24-bit tests; beyond the resolution of the E-MU).

PCM 16/44:

The most common digital sampling rate is of course "good" old 16/44 Red Book format. These days, any DAC worth it's salt MUST perform close to ideal at 16/44...

Summary (RightMark 6.2.5):

As you can see, the UD-501 was measured with the 3 digital filters (DF's - OFF, SLOW, SHARP). Comparison was made with the Logitech Transporter (ethernet), Touch (ethernet), and Oppo BDP-105 Blu-Ray player (USB). Unless explicitly adjusted, vast majority of DAC's utilize some form of the SHARP filter by default at least for 16/44. Overall, you see that 16-bit audio is absolutely no problem for any of these devices.

Here's the frequency response of the devices:

The most obvious thing to see here is that the Touch rolls off on the low end by 1dB compared to the others, and the digital filter "OFF" setting of the UD-501 rolls off quite early starting around 5-6kHz...  Let us focus for a second just on the TEAC:

Interesting; the SLOW and SHARP settings are pretty self explanatory (the Transporter has similar settings). The OFF setting results in significant roll off even earlier and by about 18kHz, the OFF setting is about -2.4dB, compared to SLOW setting at -1.7dB, and SHARP at -0.1dB.

Hmmm, where have we seen that kind of filter "OFF" curve before? Oh yeah, the old TDA1543 :-). Here ya go:

What does this mean? Yup, the TEAC can function in "NOS [NonOverSampling] mode" with the digital interpolation filter turned off! Behold, stair-stepped NOS waveforms out of a modern DAC with DSD capabilities:

Digital filter OFF:

Digital filter SHARP:

That's really quite a trick from the TEAC!

Noise Level:

All pretty equivalent with fantastic level of functioning. Note the 60Hz powerline hum visible.

Stereo Crosstalk:

Very close; worst being the SB Touch. Same 6' shielded RCA cable used in all tests.

PCM 24/96:

24/96 is the "sweet spot" for high-resolution PCM DAC's these days. Certainly in DAC's I have previously tested, once you go above 24/96, there's often deterioration in the dynamic range. Furthermore, I do not believe there is any scientific evidence to suggest human ears can experience sound beyond the resolution encompassed by 24/96 so it's important that a true "high resolution" DAC be able to demonstrate adequate performance at this level.

Here's the "big board" summary with a number of devices tested (you probably need to click on the image for more comfortable reading):

As you can see, for comparison I've thrown in results from the E-MU 0404USB itself, Logitech Touch, ASUS Essence One, Logitech Transporter, and Oppo BDP-105. Obviously every one of these devices is capable of >16-bit resolution showing improved dynamic range beyond the 16-bit test above. "Top tier" devices are the TEAC, Transporter, Oppo; each measuring beyond 110dB potential dynamic range using the RCA output - essentially at the limit of the E-MU's abilities (the Essence One & E-MU would also be on this list when using balanced XLR or TRS cables respectively).

Frequency Response:

Since the graph got too busy, I removed the ASUS and Oppo - they basically look like the Transporter in terms of frequency response. Again, the Touch drops a dB down near 20kHz.

Here's the graph with just the TEAC settings plus the old TDA1543 NOS:

Notice again the similarity of the TEAC's digital filter "OFF" setting and the TDA1543 NOS DAC in the high-frequency end.

THD:

Interesting increase in high frequency noise with the digital filter "OFF". Looks like unfiltered delta-sigma noise shaping coming through?

PCM 24/192:

Next, one more step up in sampling rate:

For interest, I threw in the Logitech Touch with EDO plugin --> coaxial --> AUNE X1 DAC. Notice how well this combination measures! The AUNE X1 is only a $200 DAC and the combination produces very respectable measurements (and sound very good IMO). In comparison, I am disappointed in the Essence One going from 24/96 to 24/192. The TEAC and Oppo really hang in there with essentially identical results compared to 24/96 - great to see!

Frequency Response:

As I mentioned in the MUSE TDA1543 measurements, one way to improve NOS DAC performance is to feed it with higher sampling rate data...  In doing so, you get closer to the performance of oversampling interpolation filters. You see this here - the higher the sampling rate, the closer the digital filter "OFF" curve gets to the "SLOW" and "SHARP" settings (in fact, you see in the next section, they become identical).

There's that early roll off with the ASUS Essence One previously measured.

Noise Level:

Essence One getting a bit noisy at high sample rate compared to the others (realize it still has >100dB dynamic range though). Again, we see quite a bit of high frequency noise with the digital filter "OFF".

PCM 24/384 (more than DXD [352.8 kHz]!):

This is a "pseudo-test" actually. The fact is that the E-MU 0404USB is incapable of digitizing at 384kHz so what I did was upsample the 24/192 test signal using SoX so see if running the TEAC at the higher sampling rate will cause a measurable loss in the analogue output dynamic range or worsen noise characteristics within the measurable capability of the E-MU.

Summary:

First 3 columns were measurements done with the TEAC running at 24/384 with various digital filter settings. The last column is the "SHARP" filter measured at 192kHz. There may have been very subtle loss in dynamic range. Some or even all of this could be due to the upsampling conversion algorithm. In any case, the measurements look excellent and it seems indeed the TEAC is able to maintain low noise even at the extreme sampling rate of 24-bit & 384kHz!

Frequency Response:

Note how the NOS-like digital filter "OFF" setting is identical to the other settings now. Basically, sampling at 384kHz is like 8x oversampling of a 44kHz signal (2x oversampling of 192kHz).

Jitter:

As usual, let us look at some FFT's from the Dunn J-Test. For simplicity, I'll just show the spectra from the SHARP filter setting.

USB input (16-bit and 24-bit spectra):

Coaxial input using CM6631A USB to S/PDIF:

TosLink input using CM6631A USB to S/PDIF:

TosLink input again fed by CM6631A with *24/192 upsampling*:

The reason I didn't bother showing any results from hardware upsampling to 24/192 in the tables above was because the numbers and graphs looked essentially unchanged. However, there is one situation where upsampling makes sense... The same reason Benchmark chose to use ASRC (Asynchronous Sample Rate Conversion) for the DAC1 and DAC2 - jitter reduction. Although by no means high, the sidebands are more pronounced using coaxial and TosLink interfaces. The sideband peaks around the primary signal clearly were reduced with 24/192 upsampling using the TosLink input. As usual, whether anyone can actually hear this difference in properly controlled testing is another matter!

Summary of PCM Results:

TEAC has created a machine which objectively compares very well to some other excellently measuring devices like the Logitech Transporter and Oppo BDP-105. It's great to see that even operating at the extremely high DXD-level sampling rates, noise level remains low and dynamic rage appears preserved.

What I found surprising was the option to allow the digital filter to be turned "OFF"; I don't recall any reviewers spending much time on this (even the AudioStream review just glossed through this and didn't comment on the sound). This setting puts the DAC into a "NOS mode" where digital interpolation is suspended - this appears novel especially in a device with low-jitter asynchronous USB interface and a true 24-bit (err... ok, 32-bit as if that makes a difference) DAC... In general NOS DACs these days are still based on obsolete decades-old DAC chips like the Philips TDA154x (16-bit) or Analog Devices AD1865 (18-bit) which tend to perform poorly on measurements. Although personally I am not a big fan of the roll-off and aliasing distortion, some have commented on subjective improvement by taking out the digital oversampling filter, so I definitely consider it a positive that TEAC offers this option for anyone to try (in real time with instantaneous A-B'ing no less just by turning the knob)! I can certainly see this option useful to tone down some of the overly "bright" digititis-inducing recordings. Looking at my pop CD collection, an example where this was demonstrable was Jason Donovan's disco-inspired Too Many Broken Hearts from Ten Good Reasons (first pressing, 1989) where the OFF setting was more tolerable after 3 minutes :-). As a compromise, the SLOW filter may be reasonable.

As I mentioned at the beginning, PCM remains the cornerstone of digital audio. These TEAC UD-501 results suggest that nothing has been sacrificed in terms of performance in the PCM domain. Note that the ASUS Essence One is also based on the PCM1795 chip in dual-mono configuration but doesn't measure as well, highlighting the importance of the electronics around it like the analogue output stage, power supply and USB/coaxial/TosLink interface circuitry affecting the final output quality. One thing I wish the TEAC had from the ASUS is the beefier headphone amp though.

Bottom line: these results are consistent with the excellent subjective sound quality described in the previous UD-501 blog post. I would happily present some kind of award if it meant anything :-).

MEASUREMENTS: TEAC UD-501 DSD Performance (Part 3)

May 12, 2013, 8:18 pm

≫ Next: PROTOCOL: [UPDATED] The DiffMaker Audio Composite (DMAC) Test.

≪ Previous: MEASUREMENTS: TEAC UD-501 PCM Performance (Part 2)

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Okay, this is the 3rd (and likely) last part of my TEAC UD-501 review.

This will be the second time I measure a true DSD device. The first time was about a month ago when I checked out the beta firmware for the Oppo BDP-105 at a friend's house. Unfortunately, although we seemed to be able to play some DSD128 samples from 2L properly in DFF format, I was unable to play the DSD128 encoded test signals properly, so this will be the first time on this blog I can show the improvements between DSD64 and DSD128.

Setup:

The setup is the same as with my PCM tests.

MacBook Pro --> shielded USB --> TEAC UD-501 --> 6' shielded RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

DSD playback software: JRiver Mac alpha 18.0.177.

I used the same KORG AudioGate 2.3.1 DSD-encoded test signals as I did previously with the Oppo tests. Basically, I took the highest resolution 24/192 synthetic test signal produced by RightMark 6.2.5, ran them through AudioGate to produce the equivalent DSD64 and DSD128 versions to test. Of course, doing this involves a transcoding step so the result at best is that of the PCM signal minus small losses due to the transcoding using AudioGate. These tests will not demonstrate the best that the TEAC (or whichever DSD device tested) can do, but rather, hopefully a reasonable approximation. Doing this also implies the possibility that different conversion software could produce different results.

As you likely are aware, 1-bit quantization results it lots of noise. DSD deals with this by virtue of the high sample rate - DSD64 at 2.8MHz and DSD128 at 5.6MHz along with noise shaping to shift the noise up into the ultrasonic parts of the spectrum. By doing this, DSD is capable of high signal-to-noise ratio in the audible spectrum rivaling that of 24-bit PCM. However, this noise floor is not flat like for PCM, but rather will gradually increase higher up in the spectrum and then escalate rather quickly by 20kHz for DSD64.

As I showed in the Oppo test, when you look at something simple like a 1kHz sine wave through DSD64, you can actually make out this high-frequency noise. Here is how it looks coming out of the RCA output from the TEAC:

DSD64 1kHz sine wave at -6dBFS:

DSD128 1kHz sine wave at -6dBFS:

PCM 16/44 SHARP digital filter, 1kHz sine wave:

 

As you can see, the DSD128 and PCM (with digital interpolation filter) waveforms look nice and smooth compared to the DSD64 output. Good "concrete" example of the improvement with DSD128.

As I mentioned, noise shaping will move the excess noise up into the ultrasonic spectrum. Due to concerns that this ultrasonic signal may cause some amps to oscillate, an analogue lowpass filter is used to remove some of this noise. The finite impulse response (FIR) filter is selectable for the TEAC based on the options in the PCM1795 datasheet:

FIR1: fc = 185kHz, gain = -6.6dB

FIR2: fc = 90kHz, gain = +0.3dB (default)

FIR3: fc = 85kHz, gain = -1.5dB

FIR4: fc = 94kHz, gain = -3.3dB

Now before one gets overly impressed, realize that these filters operate in the ultrasonic range... That is, you're not going to hear the difference other than the potential gains or attenuation as it may affect the audible frequencies. However, those gain values are very audible indeed with FIR2 sounding clearly loudest and FIR1 softest (again, like for the PCM filters, just a turn of the knob allows you to instantaneously A-B the difference on the TEAC).

DSD64 (1-bit, 2.8MHz):

Let's get to it then, RightMark 6.2.5 results playing the DSD64 transcoded 24/192 test tone:

The first 4 columns are the TEAC with FIR filters 1-4. The fifth is the TEAC playing the original PCM test with the SHARP filter (best measuring of the three). Finally the last column is the Oppo BDP-105 playing from a USB stick.

Notice the default FIR2 filter performed the best.

Here's what the frequency response looks like:

Notice how little difference there is between the 4 FIR filters! I believe this is to be expected since the analogue filters as I mentioned are operating way in the ultrasonic spectrum and differences are seen only above 20kHz. In comparison, the PCM test behaves appropriately (yellow), and the Oppo (red) interestingly has a filter that measurably starts rolling off by 10kHz (no biggie, still only down by -0.7dB at 20kHz).

Noise Level:

THD:

It's quite evident from the graphs above just how noisy DSD64 is above 20kHz. The 24/192 PCM test signal (yellow) is cleaner beyond 20kHz. The noise characteristics of the TEAC and Oppo are about the same with a hint that the Oppo's analogue lowpass filter is stronger.

DSD128 (1-bit, 5.6MHz):

Finally, I get to have a look at what DSD128 measures like. Theoretically, 1-bit sampling going from 64x to 128x should allow the machine to push the noise an octave higher. Therefore, we should see the noise start ascending from 40kHz onwards. Likewise, the SNR in the audio spectrum should improve as well but this would be already beyond the E-MU's ability to measure.

First, the Summary:

Hmmm, nice. No deterioration in noise or dynamic range compared to DSD64 (like I said, these should be even better at DSD128 but will need a better measurement device). Again, we see the fluctuations in measured values between FIR1-4. The 5th column again is the native PCM 24/192 SHARP filter result.

Frequency Response:

Interesting, once we hit DSD128, they're all essentially identical. This too could be just a limitation of the E-MU hardware and RightMark software.

Noise:

THD:

In both the graphs above, we see that DSD128 indeed does push the noise floor out to ~40kHz as predicted and from there, the level rises quite significantly. Although the noise level isn't as high as DSD64 by 100kHz, it is still significantly higher than with PCM (yellow).

Dunn J-Test:

As I've mentioned before with the Oppo tests, you cannot apply the J-Test to DSD and expect to stimulate jitter since this was designed for PCM with 16/44kHz or 24/48kHz sampling rates in mind; specifically for digital S/PDIF or AES/EBU interfaces between transport and DAC. Nonetheless, here's what they look like through the TEAC in DSD64 (DSD128 looks about the same).

16-bit J-Test:

24-bit J-Test:

Looks perfect (as it should with DSD).

Summary:

Well everyone, there you have it. DSD64 and DSD128 off the TEAC UD-501. Contrary to the French Qobuz review where the authors have suggested that DSD was being converted to PCM internally, the test results here are consistent with direct DSD decoding. For a comparison, look at the results with the Pioneer DV-588A where internally 24/88 PCM conversion was being done on DSD64 (look at the unusual noise spectrum, PCM-type frequency response, and J-Test showing the jitter pattern found in PCM).

The TEAC UD-501 performed essentially the same as the Oppo BDP-105 from what I see here. This is good and provides a nice comparison with the level of performance out of the SABRE32 DAC. Again, this level of performance certainly is consistent with the subjective listening of DSD64 and DSD128 material I reported earlier.

About those FIR filter settings. There really is no difference in sound other than the differences in gain from my listening. I'd just stick with FIR2 unless there's a reason you would want to attenuate the signal (for example, FIR1 with -6.6dB allowed me to measure the TEAC through the XLR output without clipping the E-MU 0404USB - results are good BTW and shows the benefits of balanced interconnects).

Considering the ultrasonic characteristics of these filters for a moment, FIR1 with a cutoff frequency (fc) of 185kHz basically will allow all the DSD noise to pass through unattenuated. So, if you figure you have an amplifier that can handle all that ultrasonic noise feel free to give this setting a try... It's like the old "custom" filter setting of the first Sony SCD-1 (vs. the stronger "standard" filter) where some audiophiles felt the sound became more "open" and "airy" with the weaker filter. FIR3 is the strongest filter out of the 4. Personally I'm happy with FIR2 with its fc of 90kHz. Look at the PCM1795 datasheet for some nice graphs for these filters.

Benefits of DSD128 over DSD64 are clear in the measurements - you can see it in the cleaner sine wave above, and it pushes the ultrasonic noise out to ~40kHz. Subjectively, it clearly has the potential to sound fantastic if the recording is up to par. I guess we'll see in the days ahead just how much commercially available material gets released...

Bottom line: For the price, the TEAC UD-501 DAC offers up a lot of value and fantastic set of features. Sonically, I believe this DAC can easily trade punches with the best out there - whether PCM or DSD.

Enough with testing... Time to enjoy the music!

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NB:

Before I end off, I just want to make a general comment & plea about the state of DSD computer audio now that I've got a chance to try it out.

The DFF and DSF file formats are inadequate. Compared to FLAC, APE, or ALAC, these DSD file formats feel geriatric! Seriously, PLEASE get the file format right.

Firstly, we need good tagging features - all the more important for DSD since much of the excellent material consists of classical music where it's important to document conductor, orchestra, composers, title, year of performance and composition, etc... Please let me be able to use something universal like the excellent Mp3tag to manage all my PCM and DSD files.

Secondly, a standard DSD file format NEEDS lossless compression. DSD is extremely compressible - using DST with DFF files, I regularly see compression ratios >2.5:1 losslessly, getting up to 3:1 in some tracks. This becomes even more useful for DSD128 where the space savings are very substantial. By doing this, DSD64 can be compressed to file sizes overall smaller than 24/88 encoded with FLAC with equivalent (some would say better) sound quality... I'd certainly be happy with that! Lossless compression would also save file transfer times and cost of storage for the music producer, distributor, and of course consumer. Seriously, what other modern hi-resolution media format doesn't allow for at least lossless compression?

Over the months, I have heard DSD apologists talk about how you don't need compression or native tagging because "hard drives are cheap" and "JRiver can tag with its internal database". Sorry... That's not good enough. This is a foundational matter and will impact future generations of products, so it's important to get it done properly instead of rely on work-arounds.

Please guys, now that you've gotten together to define DoP and manufacturers to make these DAC's, lets get it done right with a standard DSD format that's fully capable, preferably "free" as in "open". Maybe some enterprising coder like Josh Coalson of FLAC fame can apply their expertise!

Thanks in advance. ;-)

PROTOCOL: [UPDATED] The DiffMaker Audio Composite (DMAC) Test.

May 20, 2013, 4:09 pm

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

≪ Previous: MEASUREMENTS: TEAC UD-501 DSD Performance (Part 3)

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Up to now, I have been using primarily a combination of RightMark along with the Dunn J-Test for my audio measurements. IMO, the standard procedure I've used thus far isn't bad and can already detect many anomalies in the hardware tested so far within the limits of the test system (ie. using my E-MU 0404USB as the ADC).

In the days ahead, I am going to start doing some Audio DiffMaker tests where appropriate; another freely available tool for the audiophile tester to find out what works, what doesn't, and to identify the difference. If you have not already guessed, some of my motivation in doing these tests is not only to feed my own curiosity, but also to encourage others to understand the tests and technology - hopefully in time elevate the knowledge base rather than unquestioned acceptance of many senseless audiophile myths out there.

If you peruse the DiffMaker site, it's quite obvious what this program does. It basically takes two recordings of the audio (presumably under 2 conditions or with different hardware), inverts one of them, and applies it to the other to see if the signals "null" each other out. The "magic" of course is in the algorithm used to align the samples in terms of time (including sample rate drift), and signal amplitude. If the recordings are identical, there should be a complete null where the result is silence. The program will create the "null" WAV file to review (very useful) and spit out a number representing the amount of "audio energy" left in the resulting null'ed audio file - expressed as dB's. The program calls this the "Correlated Null Depth". The higher this value, the more correlated the 2 samples are (ie. the "closer" they sound).

The beauty of this method is that one is free to use any audio input signal - freed from the need to remain bound to synthetic test tones which thus far I have been using. The main limitation so far with this software I have seen appears to be memory limits I've run into with long audio segments, it also takes a fair bit of computation to get the results. With my 6GB Windows 8 x64 laptop and DiffMaker 3.22 (September 2008), once I go beyond ~35 seconds 24/96 audio, the program runs into an error condition - presumably memory issues. Fair enough, I think 35 seconds is adequate to allow a decent comparison.

After a bit of consideration, I decided to create a "composite" audio test signal that I hope represents a reasonable survey of real music that is also challenging enough for a high-end audio system to reproduce.  For fun, I've called this audio track the "DiffMaker Audio Composite" (DMAC) Test which I think would be a reasonable test to apply to future evaluations I post on the blog. The DMAC consists of the following 4 tracks - all downsampled to 24/44kHz. Why you may ask? Simply because most digital music exists as 44kHz so it's important that this sampling rate be done right, and it is believed by many that 24-bit depth is the major factor lending improvement to hi-res audio quality. The tracks:

Rebecca Pidgeon - "Spanish Harlem" 3:02-3:11 (The Raven, 1994) - 9 seconds taken from the 2009 Bob Katz 15th Anniversary Edition at 24/88. Well known to most audiophiles as a vocal test track... Shakers in the background and such... Good evaluation of the mids.

The Prodigy - "Smack My Bitch Up" 2:13-2:22 (Fat Of The Land, 1997) - 9 seconds of loud and clipped techno/electronica. I applied -2dB to the track to allow extra headroom for the ADC without clipping. Low dynamic range, but intense bass. An example of "modern" mastering efforts. Taken from the CD 16/44.

Rachel Podger & Brecon Baroque - "Concerto In G Minor, BWV 1056: Presto" 00:02-0:10 (J.S. Bach: Violin Concertos, 2010, Channel Classics SACD to 24/88) - 8 seconds of lovely string classical work - good mid-range to highs, nice "microdynamics".

Pink Floyd - "Time" 00:06-00:10 (Dark Side Of The Moon, 1973) - 4 seconds of bells & chimes taken from the start of this track. Quite a lot of high-frequency content, detail in the sound, and channel separation. I used the 2011 24/96 Immersion Box Set remaster.

Interspersed between each track are dual bursts of 0.1s 1kHz tone at -4dBFS interspersed with 0.1s silence. This serves as a "beacon" for DiffMaker's alignment algorithm. The trickiest part of this test is temporal alignment and doing this has significantly improved the consistency of the results for me.

DMAC Waveform:

Vital stats for the 35 second test track:
DR9 (thanks in a large part to the loud compressed Prodigy track). Peak volume: -1.37 / -1.46 dB. Average RMS Power: -27.1 / -26.66 dB.

As with any proposed test, first thing to do is some form of validation.

I. Reliability

Setup:MacBook Pro Decibel --> shielded USB --> TEAC UD-501 (SHARP filter) --> shielded  RCA --> E-MU 0404USB --> shielded USB --> Win8 laptop

Although the DMAC track is 16/44, it was measured back at 24/96 where the E-MU 0404USB functioned optimally. I also turned ON compensation for sample rate drift. The rest of the settings are as per default.

Here are 15 runs with the DMAC track played back through my TEAC UD-501 looking at the reported "correlated null depth" as an objective measure by the program. I also had a look at the null waveforms to ensure there were no obvious technical issues. The runs were spaced out over 24-hours to capture changes in conditions that may be present over the course of the day, temperature variation, electrical condition, and how long the DAC and ADC had been turned on in order to get a sense of the error range. Interestingly, from what I can tell, the result seemed to vary with ambient temperature. Trials 4-8 were done in mid-day with temperatures going up to ~30 degrees Celsius where I did the tests. Of course, maybe other factors like electrical noise and powerline quality may have a hand in the variation during that time of the day. In general, since I do most of my testing in the evenings, those lower results serve as a reasonable lower extreme for this test. (BTW: I turned the WiFi off on the computers if anyone thinks that makes a difference.)

As you see, there is a range of results (mean = 80.74/79.66, standard dev = 3.88 / 3.89). Remember that because we are measuring the analogue output from the DAC, there will be some noise in the signal - this is an inevitable property of analogue signals especially since I'm re-digitizing it back with the ADC to measure.

II. Validity

Given the error range above, is it good enough to detect very small changes?

Let's try to measure the following conditions:

1. Adobe Audition 3 Graphic EQ boost of +0.3dB at 16kHz with another EQ boost of +0.3dB at 5kHz. The 16kHz change should be inaudible, and the 5kHz adjustment likewise should be inaudible except maybe to the best young golden ears. I was unable to ABX this EQ change using the Sennheiser HD800 + TEAC UD-501.

2. TEAC UD-501 digital filter set to SLOW. This involves a high frequency roll-off starting north of 15kHz. May be detectable to those with excellent high-frequency hearing but I think for the vast majority of us, this difference is unlikely to pass an ABX test.

3. TEAC UD-501 digital filter set to OFF. This is of course the "NOS" mode for the TEAC. I can quite readily hear the difference in an A-B test. Should not be a problem for the DMAC protocol.

Reminder of the TEAC filter frequency response curves:

Result of test conditions 1-3:

Not bad. Note that I only did 5 runs of each test condition (vs. 15 runs for the DMAC Reference). The Graphic EQ test and especially the "NOS" mode demonstrated significant variance from the Reference results. Setting the digital filter to SLOW hinted at lower correlation depth but remained within the range for the Reference tests suggesting that the DMAC protocol was unable to differentiate this condition (not surprising by the way since musical content drops off significantly up at 15+kHz where the SLOW roll-off operates).

4. Changes due to MP3 encoding. We know lossy encoding changes the bit-perfect nature of the signal. We know ~320kbps is audibly very subtle (as per the test that kicked off this blog). We know that lower bit rates will result  in more sonic degradation. Can the DMAC test differentiate MP3 from the lossless and further discriminate different bit rates using LAME 3.99.5 (3 runs each condition, CBR="Constant Bit Rate")?

Nice, it looks like indeed we can! Good correlation between decrease in "correlated null depth" (increasing variance) and lower bitrate for MP3 encoding. The machine isn't fooled by MP3 algorithms :-).

Of course there are other things I can do to demonstrate the validity of this test to show variance... I've done a few other things like varying degrees of EQ changes to demonstrate the correlation which I won't bore you with here.

Summary:

As you can see, it looks like the DMAC Test is quite reliable and can be shown to discriminate differences in audio even down to levels that are very unlikely to be heard by human listeners with the E-MU 0404USB as a measurement device.

A word about tests like this and audibility. Remember that humans listen with a powerful psychoacoustic "filter". The ear has significant physiological limitations. For example, we are sensitive especially to the 1-5kHz audio spectrum and quickly lose sensitivity to frequencies higher up - have a look at the Fletcher-Munson curves. Secondly, psychoacoustic effects like simultaneous and temporal masking renders certain details inaudible. This is part of the "magic" of lossy encoding algorithms - allowing software to throw out quite a lot of data/details yet maintaining excellent audio quality. (Interestingly, the DiffMaker program does have an "ARM-468 weighted energy" setting which may be closer to human perception but I have thus far not tried it yet.)

The results of tests like this one I believe can be used for correlation of the sonic output to demonstrate variance between signals (which is of course the intent of the software developers). However, because the machine does not have the psychoacoustic mechanism of humans, the results can never directly correlate with what is being heard subjectively. A good example is the similar score between the digital filter OFF (NOS) condition and MP3 192kbps. They both score around 50dB in "correlated null depth", but I would argue the MP3 encoding changes the sound significantly less than removing the digital filter (ie. the effect from a NOS DAC). In an AB test, I can detect a "dulling" of the high frequencies on tracks like the Prodigy sample with the digital filter turned off whereas the MP3 sounds less 'colored'.

One more thing about using the "Correlated Null Depth" value. What I'm showing here is all based on the measurements off my equipment using the E-MU 0404USB, TEAC UD-501 DAC, and procedure/settings I'm using. This means it's only useful for my test purposes and cannot be generalized otherwise. The measured value itself of course will fluctuate and time-to-time, I'm going to need to readjust the reference score based on hardware changes.

I look forward to incorporating this test with the others in the days ahead...

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Addendum: Curious to see the difference between Reference null and what happens without a digital filter (ie. "NOS mode" on the TEAC)?

The following is what a high quality null WAV output looks like (~85dB) - "Spectral Frequency View" where the X-axis is time and Y-axis is frequency with the color representing amplitude at that specific frequency (blue/dark = low amount, red/bright = high):

Here's the TEAC UD-501 in "NOS mode" with digital filter turned off:

Impressive amount of variance. Also note the amount of high frequency content being recorded above 20kHz without the filter in place!

UPDATE: June 4, 2013
As requested, I've posted the DMAC test file as described above for download. Remember, this is like the "snips" of audio I posted months ago for the MP3 test. Although copyrighted material has been used to construct this test file, I believe it is being utilized as "fair use" for the purpose of education / demonstration...

 http://filepost.com/files/6mfe4c74/Archimago_-_DMAC_(24-44).zip

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

May 22, 2013, 8:40 am

≫ Next: MEASUREMENTS: Bit-Perfect Audiophile Music Players (Mac OS X).

≪ Previous: PROTOCOL: [UPDATED] The DiffMaker Audio Composite (DMAC) Test.

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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: