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MEASUREMENTS: WD TV Live - A look at (and listen to) the digital "low end".

August 11, 2013, 8:59 am
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I'm sure we've all seen these ubiquitous devices at the local BestBuy, Costco, Walmart, etc.



Even as an "audiophile", you might be tempted to purchase one for streaming audio/video to the den, basement, bedroom, etc... As you can see, on the back, from right to left, there's the little phono plug which functions as composite video/stereo audio (comes with a supplied cable), then a second USB port (one up front), HDMI, 10/100 ethernet, TosLink, and power plug to the wallwart.

It's mass market, inexpensive (this particular no-frills model usually goes for <$100), and there are a number of equivalent digital streamers out there with a similar feature set (Patriot, Roku, D-Link, Pivos, etc.). Although it's primarily meant to be a digital HDMI transport, I wanted to see what this could offer for the audio lover... In my mind, "mass market" and "inexpensive" are not bad characteristics. IMO consumers should be thrilled to find a technically good/excellent product at this price point and ease of availability if they can! Furthermore, I think it's worth looking at the "low end" to understand just what is gained with better quality gear more to the "high end".

The P/N at the bottom of the unit was WDBHG70000NBK-01. This was bought back in 2011 by my brother-in-law so current models may have hardware differences. I tried to open the unit up to have a look inside but there are no screws and I really did not want to potentially damage the aesthetics (it's not mine after all!). I see Legit Reviews opened an even earlier unit back in 2009 and found a Sigma SMP8655 SoC inside but didn't comment on what DAC was being used. I imagine this model would be based on something similar. Legit Reviews has another article on this model but no discussion on the chipset.

Let's see objectively then what this little device can do...

I. Stereo analogue output:

Let us first start with looking at what's coming out of that composite/stereo RCA cable from the built-in DAC. The supplied phono-to-RCA cable is cheap and thin but functional, about 3 feet long.

Setup:
Test signals & music (FLAC) on high speed ADATA USB3 stick --> WD TV (front USB) --> supplied audio/video RCA cable --> E-MU 0404USB --> shielded USB --> Win8 test laptop

WD TV firmware (latest): 1.16.13

I'm using a high speed 32GB USB3 stick. Although it's a USB2 port, I did not run into any troubles. All audio was encoded in FLAC lossless compression. This setup should provide the best audio quality from the unit (ie. no streaming issues or risk of lossy transcoding). There's no digital volume control to affect the sound quality as far as I can tell. I mainly want to see just how objectively accurate this device can decode audio through its own DAC and as an optical TosLink transport later.

Here's the 0dBFS 1kHz square wave through the oscilloscope:
Good channel balance and credible square wave. Peak voltage at ~1.23V. Notice the plateau isn't flat suggesting imperfections in the voltage regulation.

Impulse response:

Somewhat unexpected, looks like an intermediate phase upsampling digital filter was used which decreases the pre-ringing for a more extended post-ring. Absolute polarity maintained.

RightMark:
Here is a summary of the "big board" - I've included data from the Squeezebox Touch and with the Transporter representing the kind of result one usually associates with "high end" products:

Basically we see that this device is capable of 16-bit resolution with marginal improvement going to 24-bit data. Interestingly, it seems to be handling 88kHz okay - good frequency extension beyond 30kHz:


But, 96kHz and above (I also tested 192kHz) looks like it's being downsampled to 48kHz (verified when I did the digital tests below):

Odd, I wonder why they didn't support 96kHz since it's not that much higher than 88kHz and arguably more useful.

A few comparison charts at 24/48 then:
Frequency Response:

Noise level:

THD:

Clearly the WD TV Live cannot compete with either Squeezebox products technically. Frequency response is down to -3dB by 20Hz which can result in audibly weak bass. Also, there's some kind of high frequency noise at around 16kHz.

Jitter:
As I have demonstrated in the past, jitter (as can be measured with the Dunn J-Test) is usually NOT an issue unless there's an S/PDIF interface in the way.  This is true with the WD TV Live:

16-bit Dunn J-Test:

24-bit Dunn J-Test:
No anomalous sidebands with the J-Test at all. No surprise... However, that nasty 16 kHz noise can be seen in the graphs above!

1kHz -90.3dB Waveforms:
So, what does the 1kHz -90dB undithered 16-bit waveform look like?
Hmmm. As you can see the left (green) channel is very noisy compared to the right (blue). In fact, you can generally make out the 3 voltage levels in the blue tracing suggesting good representation of a 16-bit bit-perfect signal. I determined that the 16kHz noise was primarily in that right channel...  Ugly, but arguably still better than the TDA1543 NOS DAC I showed in the previous post. Remember that this is zoomed in looking at a -90dBFS signal.

Here's the same waveform in 24-bits:
Again, noisy left (green) channel with a smoother sine wave with the right (blue) tracing.

Okay... Obviously the analogue output leaves much to be desired straight out of the WD TV Live.


II. As S/PDIF TosLink Digital Transport:

Setup:
Test signals & music (FLAC) on high speed ADATA USB3 stick --> WD TV (front USB) --> 9' generic plastic TosLink --> ASUS Essence One DAC --> 3' shielded RCA --> E-MU 0404USB --> shielded USB --> Win8 test laptop

I tested the WD TV Live with audio set as "Stereo" as well as "Digital TosLink Pass Through" in the Setup menu and noticed no difference for regular PCM audio.

RightMark:
The results are clearly improved over the analogue output above. Basically, these numbers are in line with the usual result out of the Essence One using unbalanced RCA cables. (Note that some of my other tests are with XLR balanced cables which usually improves dynamic range by about 3dB.)

You might be curious why there's no 24/88 result...  Interestingly, even though 24/88 could be played with the analogue output, it doesn't output a digital signal at that sample rate! I can see the Essence One going into 24/88 mode but there's only silence! I don't believe this is an issue with the DAC since with the Squeezebox Touch (EDO kernel), I am able to play up to 24/192 using the TosLink interface (most device pairs are limited to 24/96 with TosLink).

Again, 24/96 is downsampled to 24/48:

Let's now compare the 24/48 result with some of  the other transport devices I used in my previous post comparing various digital transports connected to the ASUS DAC via TosLink:

Frequency response:

Notice the slight variability between the devices up in the high frequency range.  Again, my suspicion is that this is due to slight timing differences in the S/PDIF signal. Zoomed in, you see that the WD TV Live is actually right at the middle of the pack: 

Noise Level:

THD:

Stereo Crosstalk:

As you can see, other than that slight frequency response difference, the other tests show no significant difference between the digital transports with the ASUS Essence One DAC. Unless you have better than 0.1dB hearing acuity up at 18kHz, that slight frequency response variance between the devices should not be significant.

Jitter:
16-bit Dunn J-Test:

24-bit Dunn J-Test:

Well, there's the S/PDIF jitter for you. Jitter modulation pattern is obvious which means we're looking at a bit-perfect signal from the WD TV. 24-bit tracing is clearly more jittery than at 16-bits.The WD TV Live's digital output is more jittery than the previous devices tested (you can find those graphs in this post).

1kHz -90dB Waveforms:
16-bit undithered:

24-bit undithered:

Much better looking zoomed-in waveforms - as expected from the Essence One DAC. Only a bit-perfect source would be able to produce that 16-bit undithered waveform morphology above.

III. Summary:

So, this is what a $100 streamer can do in terms of audio these days. On the whole, not too bad actually! Some level of inaccuracy is expected in the objective analysis; not surprising given the compromises at this price point for something that's targeted more for digital video playback/streaming.

In terms of analogue audio quality directly off the unit:
1. It's a 16-bit internal DAC that's demonstrably noisy down at the LSB level. Although it has aspirations for 24-bits, there's really no significant benefit.

2. It's curiously able to manage 88kHz but anything above gets downsampled. IMO might as well stay with 44 & 48kHz.

3. The frequency response is hampered by bass roll-off of a greater magnitude than I'd be comfortable with. This IMO is the most audible effect and results in audibly "weak" bass.

4. There are suggestions of power supply issues with the square wave stability, and electrical noise up at 16kHz especially affecting the right channel in this sample I'm testing.

5. If a person were to complain about the sound quality of the analogue output, please don't point your finger at the dreaded jitter...  The issues above are much more significant.

As a digital transport using TosLink to the ASUS Essence One DAC:
1. RightMark results off the ASUS DAC are completely in keeping with the other bit-perfect transport devices previously tested (like the Touch, Transporter, Receiver, SB3, laptop-to-CM6631A, etc...)

2. This device is strangely incapable of sending an 88kHz signal to my ASUS Essence One even though it can decode 24/88 with the analogue out. Again, 24/96 and above gets downsampled to 24/48. More reason to just stay with 44 and 48kHz sampling rates.

3. S/PDIF TosLink jitter is demonstrably elevated compared to the other devices. This is the most technically anomalous finding (apart from the limited/idiosyncratic sampling rate support).

Subjectively, I had a listen to the analogue output of the WD TV Live with my Sennheiser HD800 headphones over a couple of nights. Despite the measurable limitations, it actually doesn't sound bad. With softer tracks like Queen's Love Of My Life (DCC Remaster) and Tracy Chapman's Fast Car, it sounds reasonably detailed except for a bit of harshness in the upper frequencies ("brittle" sounding high-hats and cymbals for example on a few of the tracks). With louder/bass-heavy tracks like AC/DC's Thunderstruck or Prodigy's Smack My Bitch Up, it doesn't "rock" as hard but most of the bass is still there; just not as accentuated as I'm normally used to. On loud, compressed tracks like Tyler Bates' To Victory ('300' soundtrack), things seem congested but not unenjoyable.

Once I switched to the digital TosLink output, it sounded like the ASUS Essence One. Nice and clear, good bass definition, quiet background. Even though higher amount of jitter is demonstrated with the J-Test compared to the other devices tested, I remain unconvinced that it's audible in regular music. I agree that with a S/PDIF interface, bits are not just bits but include timing inaccuracies (jitter), however I remain unimpressed that jitter of the magnitude I'm measuring with the WD TV Live negatively affects audio quality in a meaningful fashion.

So far, I'm still of the opinion that bit-perfect digital transports sound essentially the same when connected to a decent external DAC (consistent with the results here recently and here where I tested different laptops awhile back).

Remember that for these tests, I'm just using audio stored on a USB thumb drive. I did not set up the WD TV Live to stream off the ethernet/WiFi so cannot comment on how that would sound.

Music selection tonight: Time for a little "latin jazz"? Poncho Sanchez's "Freedom Sound" (1997) and "Cambios" (1991) are great for a warm summer night :-). Well recorded, dynamic albums with sweet music...

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Guys, even though I just got back from a trip, I'm heading off to another soon :-). Busy summer with the family. Have a great August! I hope to check out some audiophile shops in Singapore this time around like The Adelphi.

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Greetings from Asia...

August 30, 2013, 4:11 am
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Thought I'd put up a quick post since it's the end of August...

Been traveling around a few countries over the last few weeks with the family. I've kept my eyes open to see if I can spot some good audio gear but so far no luck. Beijing for example has huge malls of IT gear - computers, cameras, DIY pieces, electric toys, surveillance equipment, and massive floors of flat screen TV's. Barely anything hi-fi to be found. Maybe I just didn't hit the right stores!

One thing that's quite clear these days, with the year-on-year inflation running close to 3% over the last few years, with inflation up to 8%/year back around 2006-2007, Beijing (large cities in China in general) sure isn't cheap these days for most things from a foreigner's perspective. The cost of housing/condos these days would be horribly prohibitive for middle-class young folks to set up a decent sound room.

Note that I did run into a few speakers that looked like clones of the B&W Nautilus 801 of questionable workmanship...  Otherwise, what I saw looked like quite low-end receivers and the ubiquitous soundbars meant for small home theatres.



I've been to Singapore a number of times, about every 2-3 years for personal travel and work-related duties. It's amazing the development over the last few years...  I guess opening up for gambling does tend to draw in liquidity :-). This despite decades of bans against gambling out of a strong moral stance. Behold, the Marina Bay Sands and the "supertrees" out at the Gardens By The Bay right across from it:



For tonight, I sign off from a spotty WiFi connection here in Ao Nang, the tourist village close to Krabi, Thailand. Some of the islands around here got hit pretty hard by the tsunami back in 2004.




I'll be back in Singapore next week and hope to hit The Adelphi for some hi-fi auditioning... Wishing all a good Labour Day long weekend ahead (in N. America at least) :-).


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A Visit to The Adelphi...

September 4, 2013, 4:05 am
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As promised, I managed to visit The Adelphi in Singapore to have a listen. It's quite close to the City Hall MRT (their 'subway' system), down the street from St. Andrew's Cathedral.

One day, I hope to get to check out an audio show like maybe the Rocky Mountain Audio Fest. Until then, it can be very difficult to listen to good high-end gear in North America. But this is thankfully not the case in other places like Singapore.



Over the years, I have come by to visit some of these stores whenever I've been in town (at least 3 times now, maybe more). Previously I have come with my dad, brother; this time with my 8 year old son to audition some of the gear. On a weekday, it's not busy and at most we ran into maybe another customer or two at any one time. Each time, the shop keepers have always been courteous and for the most part, they're happy with foreigners snooping around taking photos...  Good audio discussions with very knowledgeable folks as well - no pushy sales lines here and they're happy to let the music run while off doing something else. I suspect this is a better ambiance than what I hear about some of the audio shows :-). [I wish all the proprietors in Vancouver could be like this.]

Due to limited time, I only got a chance to check out a handful of the audition rooms. For the most part, they're set up very well and include room treatments like absorption panels and bass traps. As on previous trips, I brought my own CD to have a listen to some "standards"...

McIntosh / Focal Room (ONG-AV Specialists):
Focal 1028be paired with McIntosh MC601 monoblocks (600W). Nice sound. The beryllium tweeters sound fine. Good treble extension on pop tracks like Michael Jackson's "Black Or White" without harshness.


Love the analogue power meter on these things. Note the Richard Gray power conditioner in the back (upper photo). I can't remember which McIntosh CD player was being used.

Audio Note / Avantgarde Room (Audio Note Singapore):
On the other side of the equation there's also this room - very high sensitivity 104dB/W Avantgarde Uno Fino horns with low power Audio Note Quest Silver Signature 9W Class A mono amps.

Preamp was the Kondo KSL-M7, fed by an Abbingdon CD-777.

Sounds pretty good but I thought it was leaner than the McIntosh system above. Also, didn't push the volume too high... As usual, hard to evaluate systems in unfamiliar and different room setups.

Constellation Audio / Eggleston Room (Audio Note Singapore):

We've got the 250Wpc stereo Centaur amp up front. Virgo preamp mid-right of the rack, and an Audio Note CDT Five transport hooked up to an Abbingdon Digital Processor-777 DAC. Speakers are the Eggleston Fontaine Signature.

Sounds good to me. Similar to the McIntosh setup above I would say...

mbl Room (Coherence Audio):
One of my favorites over the years has been to check out the mbl showroom. I've heard the larger system here a number of times consisting of the Radialstrahler 101 E Mk II with accompanying dual 9011 as monoblocks (440W, 8 ohms), 1621A CD transport, 1611F DAC, and 6010D preamp. Without question, this is amazing sound. These speakers are all passive with sensitivity of 81dB/2.83V/m so a powerful amp is a must!

In the back in that picture above, you see the huge bass module for the 101 X-treme speakers:
Unfortunate these guys weren't hooked up.

I see on this visit they have the mbl "Corona" line of electronics hooked up with the 116F Radialstrahler speakers:
The 2 monoblocks in the lower shelf are 500W (into 4 ohms) C15 Class D amps, accompanied by the C31 CD player, and C11 preamp. Again, sounds great although I think they could use some more bass traps in that room; I noticed some irregularity in the bass line on Rebecca Pidgeon's "Spanish Harlem" for example.

Other interesting gear:
Triode TRV-88SER integrated amp. 45Wpc I believe into 8 ohms. Didn't get to hear this but very nice eye candy.


 Kondo Ongaku-Pre KSL-M77 for you boutique analogue Japanese gear lovers...

Now a couple of "omnidirectional" speakers from Duevel - the Planets and Enterprise.

Interesting looking designs. Not sure how this would sound and would love to see some measurements!

Even though I didn't get the time to peruse through the collection of goodies, there are some music shops as well carrying good collections of LP and audiophile remasterings like MFSL and Audio Fidelity...


I also had a good listen to some B&W 802 Diamonds with the Olive 6HD server thanks to the folks at Eighteen 77 but didn't get a chance to take some photos. Playing with the Olive unit, I remain impressed by the Squeezebox in terms of speed and flexibility. IMO, the Squeezebox server system remains the best I have used.

There's also a very nice headphone place which I had visited on previous trips. Nice. A good afternoon of window shopping for the boys while the girls go off to find shoes, clothes, etc. :-)

Hope this whets your appetite to visit The Adelphi for those thinking of going to Singapore.

Well, vacation time over - back to school for the kids and back to work for me :-).
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HOWTO: Getting JRiver MC19 2xDSD upsampling of PCM working on TEAC UD-501...

September 7, 2013, 11:51 am
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A quick post here for those trying to get beta JRiver MC19's 2xDSD (DSD128) upsampling working on the TEAC UD-501. [Currently I'm using beta MC19.0.37.]

See the thread here where the discussion was started - thanks for putting attention on this InflatableMouse. Apparent there are some buffer issues with the TEAC driver and native ASIO, and at this point MC19 isn't supporting a 2xDSD upsampling with DoP option. Here's a workaround:

1. Download and install ASIOProxyInstall-0.6.5 from the SourceForge link here.
2. Go into MC19 and set audio device as "foo_dsd_asio [ASIO]", Bitstreaming as "Yes (DSD over PCM (DOP))". Should look like this:
3. Now lets set up ASIOProxy itself. Go into that "Device settings..." tab. Look at the "Tools" section and click "Open Driver Control Panel...". You'll see this pop up:
Make the settings as above especially with Fs as DSD128.

4. Finally, go into "DSD & output format..." in MC19 and set output format to "2xDSD in native format":

There you go.

Should now be listening to all PCM music upsampled to DSD128 on the TEAC. Native DSD files will be bitstreamed in their respective DSD64 or DSD128 forms direct to the DAC without MC19 processing like volume control.

Using an AMD A10-5800K processor, I'm seeing CPU use peaking at ~20% and usually 10-15%  even with upsampling 24/192 music. Not bad!

Hopefully TEAC and JRiver can come up with a solution for native ASIO streaming or support of DSD128 upsampling in DoP in the future...

BTW: I just got back from holidays a little jet lagged so haven't played with this much yet. However, on my system, upsampling to DSD64/128 certainly sounds different than PCM and I can certainly see the appeal - there seems to be more weight to the bass and the sound isn't as "etched".  There is a bit of level difference between DSD & PCM playback so I will need to listen more back and forth to see which setting I prefer.

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MEASUREMENTS: PCM to DSD Upsampling Effects (JRiver MC19 Beta).

September 13, 2013, 4:26 pm
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We're continuing to see a push into the DSD domain with renewed talk of music release as digital downloads requiring the purchase of a DSD DAC to natively play (eg. recent Acoustic Sounds DSD releases). For now, I have already voiced some concerns about DSD including practical issues like the gross limitations of the file format itself. I demonstrated the noise characteristics for both DSD64 and DSD128 in my TEAC measurements. Furthermore, I have shown that there already exists many SACD's appearing essentially to be upsampled PCM from standard sampling rates of 44/48 kHz (remember, it's almost a given that any music we listen to created in the last 20 years has at some point been through a PCM stage except specified pure analogue recordings or those specifically recorded in DSD with minimal processing).

Some writers have voiced that even the process of upsampling PCM to DSD will imbue the music with some of DSD's beneficial properties, is this true? If so, what happens?

Well, thanks to ongoing advancement in the computer audio world, we can now easily have a way to listen to our PCM music as a converted DSD stream... Enter JRiver Media Center 19 and it's ability to stream PCM --> DSD64/128 in realtime to a compatible DAC. [Note that this should also be possible using ASIOProxy from foobar - not tried personally.] This will allow an easy way for everyone to listen for themselves what happens to the sound either as the original PCM or transcoded to DSD with the assurance that we're comparing "apples-to-apples" with the same mastering.

First, as has been my custom, let's start with some objective measurements to see what the DSD encoding does to test signals.

I. Objective Measurements

General Setup:
AMD A10-5800K HTPC Win8 x64 running JRiver 19.0.37 (ASIOProxy workaround for DSD128) -> shielded USB -> TEAC UD-501 -> shielded 6' RCA -> E-MU 0404USB -> shielded USB -> Win8 laptop

TEAC DAC settings:
     - PCM: "SHARP" BB PCM1795 filter, no upsampling (default)
     - DSD: FIR3 analogue filter (closest volume match to PCM output)

JRiver Media Center 19.0.37 setting:
     - ASIO buffer set to "minimum hardware size" since someone suggested it sounded better :-) - no stuttering encountered playing music.

First, let's have a look at the 1kHz SQUARE WAVE off the digital oscilloscope (24/44 source):

PCM:

DSD64 realtime conversion:

DSD128 realtime conversion:

As you can see, volume is about the same with the DSD FIR3 filter on the TEAC vs. PCM SHARP; both output ~2.85V peak with the square wave. What is also very obvious is how clean the PCM is vs. DSD. Notice the extra high frequency noise for both the DSD64 and DSD128 traces, with the DSD128 clearly less noisy. No surprise, right? If you've looked at the objective results from DSD here and elsewhere, this is pretty "normal" for DSD.

IMPULSE RESPONSE (16/44 PCM impulse):
PCM SHARP filter:

DSD64 realtime upsampling:

DSD128 realtime upsampling:

Noise is again very evident in this "zoomed in" impulse response measured at 24/192 especially from the DSD64 process. Although impulse response graphs can be excellent with MHz sampling rate (this is often a "talking point" in the DSD/Sony ad literature over the years), when resampling PCM to DSD, we're still hampered by the PCM signal's original sampling rate (eg. 44kHz). Evidently JRiver uses a typical linear phase reconstruction filter; hence the symmetrical pre- and post-ringing.

DUNN J-TEST:
PCM:
16-bit
24-bit

DSD64 realtime upsampling:
16-bit
24-bit
DSD128 realtime upsampling:
16-bit
24-bit
No meaningful differences between PCM and the DSD realtime conversion. This also means no evidence of worsened jitter with all that extra processing converting PCM to DSD in the computer at least based on the spectral output of this test (typically, this computer's AMD CPU utilization went from <5% with PCM to ~15% for DSD upsampling). DSD64 is obviously of enough resolution to accurately demonstrate the jitter modulation signal in the LSB for the 16-bit test.

RightMark:

Calculations done in the AUDIBLE SPECTRUM (20-20kHz).
Frequency Response
Noise
THD
These graphs of upsampled 24/192 test tones echo the results in the TEAC DSD Measurements back in May. I used KORG Audiogate to convert PCM to DSD back in May and it looks like the mathematical process in both JRiver and Audiogate are of similar precision. Within the audible spectrum, the PCM and DSD measurements are all very similar. A good indication of high precision in the conversion process. What is again evident is the noise once we get above 20kHz especially with DSD64.

II. Subjective Experience

As per the premise of this blog being "more objective", I'm not going to write pages on that which is experienced for oneself. However, I'll put down a few thoughts for consideration...

Listening gear:
     - Headphones: Sennheiser HD800 off TEAC DAC
     - Speaker system: TEAC UD-501 --> Simaudio Moon i3.3 --> Paradigm Signature S8 v3 (standard OFC cables)

The DSD conversion process through this TEAC DAC does change the electrical output as seen by the objective measurements above. This alone means that it's real compared to the identical measurements found with different bitperfect software and digital cables previously reported.

There does seem to be a change in the perceived detail of the sound subjectively through the gear I listed... Note that I'm taking the liberty here to not subject myself to a blind test so I fully admit that I could be wrong on this :-). Furthermore the fact that since it's not an instantaneous 'flip', echoic memory is prone to be unreliable. With these caveats, my current feeling is that both DSD64 and DSD128 conversion adds a potentially euphonic characteristic to the sound. No, IMO, it's not a dramatic difference when listening volume is controlled. [For those using the TEAC DAC, remember that the default FIR2 filter for DSD is louder than PCM by ~2.5dB - this could of course be misconstrued as sounding "better" for DSD.]

What do I hear? As I mentioned in my previous post on getting DSD128 upsampling working on the TEAC, I think the sound is less "etched". There's a pleasant subtle added smoothness to the transients. I think many may describe this as being less fatiguing, maybe less of the "digital glare". I couldn't specifically put a preference on DSD64 vs. DSD128 but knowing the ultrasonic effects, it wouldn't take much to convince me that DSD128 is better since the ultrasonic noise is further away from the audible spectrum. However, if you believe that the noise itself creates euphonia, it's also conceivable that DSD128 would sound closer to PCM than DSD64. Maybe.

I listened to a few standard 16/44 albums in DSD128 like a first pressing Michael Jackson Thriller, the well recorded Al Di Meola Winter Nights, and Suzanne Vega's Solitude Standing. They all sounded great. Like I said, marginally smoother than PCM. I think poorly recorded harsh albums may benefit even more - for example Alan Silvestri's The Avengers score is mastered in "modern" overcompressed fashion with DR9 average dynamic range (not good for an instrumental soundtrack IMO). DSD128 upsampling seemed to make it more listenable for longer duration.

Vinyl rips (24/96) of Tracy Chapman's Fast Car and Whitney Houston's One Moment In Time sounded very nice as well... "Extra" analogue from digital from vinyl :-). Again, the inability to instantaneously switch between PCM-to-DSD makes it hard to A-B compare reliably.

Unfortunately I did not take a screenshot of the phase measurements, but it looked good. Listening to phase-effect tracks such as those encoded in Q-Sound like Def Leppard's Rock On (David Essex remake off Yeah!), and Roger Waters' Too Much Rope (off Amused To Death) nicely created the impression of spatial surround and depth. Whether that sense of depth is any better with the DSD upsampling is of course debatable.

III. Conclusion

1. PCM to DSD upconversion is a DSP process. The signal output is measurably different.

2. Noise shaping pushes the DSD quantization noise into the ultrasonic frequencies as expected. In DSD64 it rises above the noise floor almost right at 20kHz, and in DSD128 it starts around 40kHz. (I vote for pushing it up to 40kHz as less likely to cause distortion through the amp & speakers.)

3. Pre- and post-ringing is similar to standard PCM with upsampling using MC19's algorithm so this would not explain any audible differences.

4. The algorithm used in JRiver MC19 does a good job with maintaining classic measurement parameters like frequency response, dynamic range, and distortion from 20-20kHz  - basically this means the math is as expected and fits the DSD output profile. Results are similar to the KORG AudioGate software converting PCM-to-DSD.

I can't help but wonder if what's happening here is like tube amps and analogue playback (eg. vinyl). Objectively the DSD conversion adds distortion but the anomalies are not perceived as objectionable and in some material, the added noise and imprecision actually makes it sound less "sterile", "clinical", more "real" (conversely being in an anechoic chamber is disturbingly unreal due to the profound silence). It would make sense to me that some people could prefer DSD64 over DSD128 upconversion since DSD64 will give you more of that distortion. Even though the noise is ultrasonic in nature as measured off the DAC, nonlinearities in the playback system like your headphones and speakers (perhaps certain amps as well) could create audible intermodulation. Maybe for certain music, this could be especially beneficial.

Out of curiosity... For anyone out there with the EMM Labs DAC2X which upsamples to DSD128, it'd be great to have a look at what the measurements are from that unit! With all the positive press about how this DAC sounds (ahem... $15.5K), I have yet to see any measurements... I wonder what a 16/44 impulse response looks like for example to see if it bears similarity to what JRiver is doing. How about the ultrasonic noise with DSD64 & analogue filter strength? Does the EMM upsampling process for PCM result in similar frequency response pattern?

In any case, give this PCM-to-DSD process a try at some point when you can. If nothing else, at least to say you've experienced it... See if you can perceive a difference and/or judge if it's beneficial for yourself.

Tonight's music: Valery Gergiev & LSO's Mahler Symphony No. 9 (2011). Nice recent classical recording available in SACD format as well.

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You might notice that I turned on AdSense on the blog.

As I have said in the past, my intention for this blog has never been about making money. I have no formal relationship with any company so have no sales incentive and am not interested in making this some kind "publication" other than what it has been - a blog about my own journey in audiophilia with a bent towards finding answers using empirical/objective means. That remains my main interest.

Nonetheless, it's trivial to "flip" the AdSense switch. I would have no idea what Google tries to market to you, and trust that the layout won't be distracting (I've switched off some questionable types of ads like for dating sites or of a sexual nature). If it gets me a few bucks for my digital downloads for what I do as a hobby anyway, I'll be happy with that!

Best regards...

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Changes... It Begins!

September 22, 2013, 3:48 pm
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As I had mentioned a few months ago in one of the responses to a post, I had plans to "upgrade" my home sometime in 2014...

As fate would have it, a house opened up for sale recently fitting the family's needs and I decided to grab the opportunity. This is going to be a very busy autumn for me and the family with a move to the new place in November! Between now and then, I've still got 2 business trips among other duties.

The upshot to the move? I'll finally have a good sized home theater space for the transition to a dedicated sound room for both stereo and multichannel listening :-). With that in mind, I've sold off the Simaudio Moon i3.3 integrated amp I had been using for the last few years...  It's time to move on to separates and the first box in this new system is this baby:

Yup, the Emotiva XSP-1 pre-amp - I got it on sale recently at 10% off (~$820USD before taxes). In the next few months, I hope to get a couple of monoblock amps for the fronts (may consider the Emotiva XPA-1 Gen 2 coming out soon). In time I'll buy a processor for the fancy "new" surround formats like DTS-HD Master Audio. For now, I figure I can live with the 10-year old Denon AVR-3802 for decode duties with the ubiquitous Dolby Digital and standard DTS - obviously multichannel isn't a major priority for now.

Just a few comments about the XSP-1. It's a nice full sized component weighing a reasonable 28lbs. The rear panel layout is good and connectors are of good quality - gold plated and robust enough to feel that they're not going to fall apart any time soon. It has a phono preamp with impedance settings which I suspect I will never use (not interested in vinyl for now at least). Line level outputs to the amplifiers available as RCA and XLR's (fully balanced topology of course is a main selling point for this preamp compared to less expensive options). I'm looking at integrating something like the Paradigm Signature SUB 1 into the system so I believe I will be using the crossover setting real soon and keeping it at the 50Hz low end. It'll easily handle a single or dual powered subwoofers. There's also a "Home Theater Input" section to easily integrate this unit in bypass mode for surround functionality.

I see that a full set of measurements have already been published by Secrets of Home Theater and High Fidelity. For curiosity, I might try out a few RightMark tests to see what the difference is between this unit and the pre-amp output from the Denon AVR-3802 at various output levels...  Could be educational.
A look at the guts... A lot of opamps in there. Under the stylized 'E' metal shielding is the power supply and resistor network volume control.
In terms of subjective sound quality, so far I'm just running this in a compromised system. With the Simaudio gone, I'm connecting the XLR from the TEAC UD-501 to the XSP-1 and routing the RCA output to the Denon's "external in" and using the receiver's amplifiers to drive the Paradigm Signature S8's... All I can say is that it does sound better than TEAC RCA to the CD input of the Denon. I wonder if the Denon is doing any internal ADC/DAC step even in the "Direct" setting which is supposed to defeat any DSP happening. For example, listening to Yello's 2009 album Touch Yello, the spatial ambiance and sense of "surround" was more prominent with the XSP-1 than directly into the Denon. Those Q-Sound albums like Madonna's Immaculate Collection, Def Leppard's Yeah!, and Sting's The Soul Cages also sounded fuller. Again, I'll see about obtaining some measurements and compare the quality of the Denon output with this new stereo preamp.

Nice metal remote. No problem getting programming for the Logitech Harmony I normally use.
If there is one criticism I have about this unit, it would be the headphone amp and output. As you can see from the first picture, the headphone jack is the small 3.5mm variety. It's fine for most of my headphones but for the higher end units like my Sennheiser HD800, I'd prefer the full 6.5mm (1/4") plug rather than having to use a converter.

The preamp will mute the line level output when it detects headphones in place so that's appropriate. The other issue is that the headphone output is relatively weak. The most power-hungry headphones I have is the AKG Q701. At full volume, it's loud enough for most rock/pop recordings. However, classical recordings with higher dynamic range and produced at softer average levels would need a more powerful headphone amp.

As always... Enjoy the music :-).

BTW: Can anyone recommend some good audio equipment stands? I'm planning to hang the flatscreen TV on the wall so don't need a TV stand. Something like the stackable Lovan Sovereigns look like a good idea since I will have enough room to put two low stacks side-by-side in front of the TV on the hardwood floor. If you know of other models/brands, I'm "all ears"!


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MUSINGS: Updates & The Value of Objective Measurements...

October 6, 2013, 11:20 pm
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Life has been busy getting things done with the new home. Also, I went ahead and bought a new Paradigm SUB 1 to add to my home theater room along with a Signature C3 center channel - piano black of course :-)...  Should be an exciting fall/winter as I get things up and running!

In the midst of this, I'm going to try getting a few comparison measurements of the pre-amp characteristics of the new Emotiva XSP-1 vs. my old Denon 3802 AV receiver to see objectively what differences can be found. There does appear to be a significant audible difference plugging a DAC into the XSP-1 then into the Denon external input and using the Denon as an amp compared to just plugging the DAC straight into the Denon as a pre-amp.

A couple of quick updates:

1. It looks like JRiver has new beta versions of JRMC 19 (19.0.51) which supports the TEAC UD-501's native ASIO DSD128 mode. No problem with using "2xDSD in native format" upsampling of PCM to DSD now. Thanks to InflatableMouse for getting TEAC and Matt over at JRiver talking. With this "fix", there's no need for ASIOProxy any more but I suppose the technique could be useful for other DAC's.

2. I continue to update the list of suspected upsampled 44 & 48kHz PCM-to-SACD titles. Thanks for the entries from various friends and E-mails over the months. Again, I think it's useful to have a look at this list if you're a collector of SACD's. Useful to ponder just what is the benefit of DSD based on these examples as well...

The Value of Objective Measurements: A Case Study


Today, I want to discuss an interesting device which I heard back in early 2012 when a friend bought one (I think he might have even been on the wait list to be one of the first to get it)... The Wadia 121 Decoding Computer.

I remember listening to it and thinking "this isn't bad". Details seemed reasonable.  The remote feels good. I like the idea of good "lossless" volume control built into the DAC. I wasn't blown away by it though. Unfortunately, this was before I started writing this blog and spending time with measurements.

Over the months since then I remember keeping an eye on what others were saying about the device. I assume Wadia did a good job sending out "loaner" units to the various audio reviewers...  A Google search shows quite a number of subjective reviews of this device. At a price point of MSRP $1299, it's not a "top end" DAC based on price but that's still quite a chunk of change. Understandably, reviewers of "high end" gear did not suggest this was the best sounding DAC they've heard, but on the whole, it received very decent, positive remarks...  Let's have a look; I'll throw up a few summary quotes in no particular order based on the reviewers' conclusions trying not to take things too far out of context: [As usual, I present these quotes as "fair use" for the purpose of discussion, criticism, and research.]

The Computer Audiophile (August 16, 2012):
"The Wadia 121 Decoding Computer is more than competent and competes with products double, triple, and quadruple its size...  New computer audiophiles seeking their first entry into this wonderful next phase of high end audio can't go wrong by starting with the 121. They may never need another digital to analog converter."

Enjoy The Music.com (August 2012):
"The Wadia 121 Decoding Computer is the best affordable digital-to-analog converter that I've ever heard. No, I have not heard every affordable DAC on the planet – and there are new DACs in all shapes, sizes, and prices being released even as we speak. But given their track record, it is a safe bet that Wadia has not only put a lot of thought into the design of the Wadia 121, but this DAC won't be bested by any DAC for quite a few years to come..."

AudioStream (June 22, 2012):
"To my way of listening, the Wadia 121Decoding Computer jumps right onto my short list of recommended components. It strikes me as being at once refined yet not overly resolute, with a voice that sounds like music. Sweet music. I enjoyed every listening minute spent regardless of the recording..."
Wins the "Greatest Bits" award.

Sound And Vision (March 14, 2013):
"...Though the music sounded like high-res digital—not vinyl—my brain still involuntarily registered surprise at the lack of clicks and pops. I suppose it associates them with a relaxed listening experience.

"Home theater buffs tend not to think much about source components for music: We figure that as long as we own an Oppo, and maybe a turntable, we’re covered. That worked as long as music streaming was a low-res medium, merely a convenient plaything for background listening. But the advent of high-res downloads demands an upgrade if you want to get the best out of your investment in components and headphones. You just may need something exactly like the Wadia 121."
Awarded 5/5 stars in the "Performance" category.

Ultra High-End Review (June 20, 2012):
"... Reading reviews is helpful (I hope), but I think a proven track record of producing high quality components is perhaps even more important. Here Wadia, well known for producing some of the finest digital playback equipment available since the earliest days of the medium, has brought its considerable talents to bear in producing a DAC which is operationally bullet-proof at an unexpectedly modest price. This is not simply another DAC-in-the-box with off-the-shelf parts and a marketing slogan, but a component with highly sophisticated software realized in DSP which has been decades in the making, coupled with an analog section which, to my ears, is completely transparent. And with no separate preamplifier needed, your budget for speakers has just doubled. I can’t recommend it highly enough."

The Absolute Sound (Feb 28, 2013):
"I can state confidently that few, if any, potential purchasers will be disappointed by the 121’s sonics or ergonomics. I know that I could happily live with the Wadia 121—it’s that good."

So, these are words of subjective reviewers. As I noted, for the price this DAC cannot really be considered "reference" level at least from the perspective of folks who likely have heard DACs in the $5000+ range and have some expectation of what these expensive DACs sound like. There are of course comments about how this DAC doesn't quite reach the level of those über-DACs. Here's a nice quote from the Computer Audiophile: "What separates the 121 Decoding Computer from the rarefied air of great but greatly expensive DACs is reduced depth, air, and low level detail when reproducing the best recordings from labels such as Linn Records, Naim, and Reference Recordings." Fair enough.

So, eventually, in the July 2013 issue of Stereophile, we get their full review. Jon Iverson's subjective comments were clearly not as positive:
"After more than a month of use and listening, when I used the 121 strictly as a DAC, I found that, in most cases, its sound had a marginally burnished or rounded quality that could help tame a recording with an unruly top end, or slightly veil a great recording."

What was somewhat stunning however was what John Atkinson found on the test bench:

"Fig.4 shows the spectrum of the 121's output while it decoded dithered 16-bit data (cyan and magenta traces) and 24-bit data (blue and red traces) representing a 1kHz tone at –90dBFS. The increase in bit depth drops the noise floor by around 9dB, implying ultimate resolution between 17 and 18 bits. To generate this graph, I fed the data to the Wadia from the Audio Precision using an AES/EBU link. To my astonishment, when I repeated the analysis using a coaxial S/PDIF link to transmit the 24-bit data, I got 16-bit resolution. The blue and red traces in fig.5 repeat the spectrum with 24-bit data and an AES/EBU link; the cyan and magenta traces in this graph were taken with the 24-bit data transmitted with the coaxial S/PDIF link. I repeated the analysis using a TosLink connection from the Audio Precision, but with no difference in the result. To check that the Audio Precision was working properly, I then used a TosLink connection from my MacBook Pro. However, I got the same result: 24-bit data but 16-bit resolution. Finally, I used a USB connection from the laptop, and although I made sure that the connection was correctly set to transmit 24-bit integer data, thenoise floor was around the 15-bit level (not shown)."(Emphasis mine.)

You can also see the noise level demonstrated in Figure 6 with the undithered -90.31dB graph. Not good. [I posted on this test back in August to show what it looks like with some of my DACs.]

Basically, what the objective results show is that we have here a fancy looking DAC with some really cool "talking points" - well respected manufacturer Wadia, "ClockLink" asynchronous USB, "DigiMaster" interpolation, 32-bit 1.4MHz upsampling. But at the end of the day, it's not capable of achieving >16-bit resolution with USB, TosLink, and coaxial inputs. Even with the AES/EBU balanced digital cable, it's "only" capable of 17-18 bits. Unless one were to just use AES/EBU, there appears to be no point feeding 24-bit high resolution audio into this DAC - all those 24-bit HDTracks/Qobuz downloads would be wasted (unless you feel >44kHz sampling is much more important). To make matters worse, it seems like the USB input cannot even achieve a full 16-bit resolution - arguably THE most important interface these days. Knowing this, how can any reviewer hand out awards or grade this device as 5/5 on performance? Even if you like the way this device "sounds", isn't it still a sign of failure that it could not profit from the higher bit depth? Of course, it appears the purely subjective reviews could not comment on this "inconvenient" piece of information.

Now, admittedly, there could have been something wrong with John Atkinson's measurements I suppose, but as of October 2013, I do not see any addendum to the review. I would imagine that a manufacturer would correct this situation ASAP!

I feel that this is a good case study (one of many IMO) into why objectivism has an important if not essential place in audiophile equipment reviews. Bias and placebo are well recognized in domains of research where human qualitative evaluation is involved. I would argue even more so when reviewing "high fidelity" gear where at a certain level of quality, differences are likely very small and effects of biases become even greater - "look and feel" (pretty metal box with lights and metal remote), manufacturer reputation (ooohhh... Wadia), price ($1299 must be a pretty decent DAC right?) all can (and likely do) end up in the final evaluation of sonic quality in the absence of objective information. In some forums / web sites, it almost seems that certain reviewers feel that they are immune to this phenomenon, or even worse, have developed so much faith in their "golden ears" that they feel there is no benefit to empirical evaluation.

Remember, thoughtfully designed audio devices are engineered. They were made based on electrical and (in instances like speakers or turntables) mechanical properties. Without examination of these properties to at least verify claims (eg. that a hi-res DAC is capable of >16-bit dynamic range being fed into it, or an amplifier is capable of the claimed watts with minimum distortion, etc...), I believe the reader cannot place strong value in the reviewer having fully appreciated the limitations/strengths of the device. I'm of course not opining that there be no subjective evaluation - fit and finish, ergonomics, ease of use, reliability, visual esthetics are all important. Likewise, sound quality needs to be checked subjectively. But there's no shame in admitting that in many ways, the human ear/brain is limited and measurement devices can easily enhance the quality of a review synergistically. There's no need to see this as black or white, subjectivist vs. objectivist.

I've said many times on the various forums how I still have a subscription to Stereophile. I look forward to reading the opinions, music reviews, and of course the gear reviews. After all that's said in the subjective portion, it's always good to study those numbers and graphs to make sure the device appears to be delivering all that was promised. I wish more magazines could do that... (Don't worry guys, I wasn't paid off by Stereophile, just wanted to give credit where credit is due.)

[BTW: Perhaps it goes without saying, I feel objective measurements are especially important with devices where the putative effects seem to be without clear scientific basis (eg. anyone know what the Synergistic "Tranquility Base" does yet?)]

Musical selection tonight:"Respect the classics, man!" -- Fillmore from Disney's Cars
     Jimi Hendrix - Band Of Gypsys

Until next time... Enjoy the tunes!

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ONKYO TX-NR1009 In The House...

October 13, 2013, 10:57 pm
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Happy Canadian Thanksgiving long weekend everyone! Sadly we don't have the same "Black Friday" deals you lucky Americans enjoy :-).

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

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

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


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

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

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

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

Onkyo tattoos!

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

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

Addendum: After writing the above about 4K last evening, I ran across this link: 4K Blu-ray is dead tech walking. Yup, sounds about right! :-)
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MEASUREMENTS: Separate vs. AV Receivers (Emotiva XSP-1 vs. Denon AVR-3802 vs. Onkyo TX-NR1009) as Analogue Preamp.

October 18, 2013, 10:41 pm
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Okay, so maybe I'm being a bit too dramatic using that epic battle between Godzilla and Rodan above :-).

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

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

First, let me introduce the contenders today:

Emotiva XSP-1:

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

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

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

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

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

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

Setup:

 

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

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

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

we get these results:

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

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

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

Results:


I. Emotiva XSP-1 RCA:

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


Frequency Response

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

II. Emotiva XSP-1 XLR:

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

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

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

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

III. Denon AVR-3802 RCA

Now we get into the AV receivers:

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

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

IV. ONKYO TX-NR1009

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


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

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

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

Frequency Response

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

Summary:

 

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

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

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

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

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


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

October 26, 2013, 10:11 am
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The rat's nest.

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

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

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

Setup:

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

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

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

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

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


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


Results:

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

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

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

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

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

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

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

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

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

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



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

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

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



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


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

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


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


Frequency Response

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

Summary:

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

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

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

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

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

Later, a more comprehensive message appeared:

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

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

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

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

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

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

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

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

November 9, 2013, 7:27 am
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By Keaton I. Goulden-Eyre III, Esq.

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

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

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The cellular phone rang -"How inconvenient!"

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

ME: And how about the ULTRA Cable?

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Objective Analysis:

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

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

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


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

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


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

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

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

Frequency Response

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



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


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

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

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

Back to Keaton for his conclusions.
 

Keaton's Konclusions:

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

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

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


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

Until next time; Magico wishes and Burmester dreams.

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* This cable is rated as "Standard HDMI".

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

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

May 24, 2013, 6:39 pm
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Do bit-perfect Mac audio players sound the same?

Over the last few years, the list of "audiophile" audio players on the Mac has gradually increased. Do they sound the same if set to bit-perfect output? Let's have a look at the candidates I'll be considering here:

1. Decibel: I bought this program more than a year ago. It's a no-nonsense program that plays a nice range of file formats without fuss. It's able to take exclusive access of the audio device, and memory playback. As with all the commercial offerings, it can switch sample rate automatically. PCM only, no DoP for DSD at this time. I upgraded to the latest version 1.2.11 for these tests. Memory playback was activated.

2. Audirvana Plus: Current version is 1.4.6. I bought this one about 6 months ago. It's got a nice, fancy GUI. Able to handle DSD files with DST and was able to play DSD64 and DSD128 over the USB interface to my TEAC UD-501 without problem. "Under the hood", it's also got some extra features like memory playback, "Direct Mode" apparently bypassing CoreAudio as well as "Integer Mode". Since the software supposedly bypasses CoreAudio, I would have thought that "Integer Mode" would be an obvious given. They also talk about 64-bit processing which is great if one has need for the SRC and dithering (iZotope-based)... For these tests, I'm using Direct, Integer Mode with memory playback to the TEAC. The green "INT" indicator turns on. Also, I have SysOptimizer turned on (disables Spotlight, Time Machine, some USB tweaks).

3. JRiver Media Center for Mac - Well known media player originating from the Windows world. I measured the beta 18.0.177 build for this test. Bit-perfect from the start so I didn't fool with any of the default settings. It's capable of DSD playback to the TEAC using DoP.

4. Pure Music - I'm not as familiar with this one. I installed the trial version 1.89g. It literally "wraps" around the iTunes interface. Can handle DSD but I didn't bother trying since it looks like there were some contortions needed to get these files recognized under iTunes. "Memory Play" was activated for playback. My subjective opinion is that I did not like the UI and using iTunes means no native FLAC support.

5. TEAC HR Audio Player - Release version 1.0 for Mac. Just a freebie I can run with the TEAC DAC. Handles FLAC. Will do DoP for DSD playback. Unable to decompress DST though. Does have an "Expand to RAM" mode which I did not use for these tests.

6. iTunes 11.0.2 - The "standard" Mac music player. Should be "bit-perfect" so long as volume at 100% and none of the DSP plug-in's are activated. A lot of uncertainly out there about this program with folks jumping up and down with each version claiming that sound has changed for better or worse... Version 11 was released in November 2012 with some folks claiming volume and sound quality changes compared to version 10. The BIG negative about iTunes for audiophiles is the lack of automatic sample rate switching - need to go into the "Audio MIDI Setup" panel to change sampling rates and bit depth (yuck). IMO, the other BIG negative about iTunes is that it does not support FLAC...  Seriously, after 11 versions, to not support the universal lossless audio format is just stupid and has been a reason why I do not buy music from Apple.

Over the years I have tried Play, Amarra, and Fidelia as well, but figure the above was enough to look at for a sense of the field out there around Mac music players. I see there's also BitPerfect for iTunes - again, FLAC limitation sucks.

Setup:

(Note that this is same as previous DMAC Test.)

MacBook Pro (*running audio 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.

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, RightMark, or jitter FFT analysis.

Part I: RightMark 6.2.5 (PCM 16/44, 24/96, and DSD64)

All the measurements done with the test signal encoded as FLAC except for those based on iTunes (iTunes, Pure Music) where AIFF was used.

PCM 16/44:

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

PCM 24/96:

Frequency Response:

Noise:

THD:


Stereo Crosstalk:

DSD64 (via DoP) - for the programs that support DSD:
Note that this is achieved using the 24/96 test signal encoded into DSD64 using KORG AudioGate, then played back to the TEAC UD-501 DAC using DSD Over PCM (DoP) protocol and measured with RightMark. As usual, we see the effect of noise shaping in DSD up in the ultrasonic range.


Frequency Response:

Noise:

THD:

Stereo Crosstalk:

Part II: Dunn J-Test for jitter (16-bits shown for brevity)

Decibel:


Audirvana Plus:

JRiver:
 

Pure Music:
Wanted to see if turning Memory Play ON / OFF had an effect.
 Memory Play OFF


Memory Play ON

TEAC:

iTunes:

No difference to see here folk... Didn't show the 24-bit test, but that was unremarkable as well.

Part III: DMAC Protocol

Time to let the machine have a listen to the music and see what kind of correlation it finds using the standard 24/44 audio sample with Audio DiffMaker... Reference for all these "correlational null depth" measurements is Decibel FLAC recording. Each audio player was measured 3 times.

I threw in comparison measurements for MP3 320kbps and 192kbps. Also to show what happens with some DSP processing - Pure Music with volume reduction of -1dB (with dither), and turned on the EQ in iTunes and dropped 8kHz slider just by 1 "click" lower.


I found it quite remarkable the drop in null depth by just turning on the iTunes EQ plug-in and using it to adjust just 1 notch (don't know how many dB's this is supposed to represent) [see addendum]! Pure Music -1dB volume control changed the measurement slightly but not much. DiffMaker has amplitude compensation so it is trying its best to compare the audio quality beyond the volume difference.

Part IV: Conclusion

Well everyone, unless I missed something obviously subtle here, what I see is that bit-perfect is indeed bit-perfect playing the audio through my TEAC UD-501 DAC with all these programs.

Now of course I cannot overgeneralize these findings to all Mac computers, all DAC's, all player programs, all drivers, all DAC's, etc... But I think I can say with some assurance given similar setups as mine that:

1. With bit-perfect playback, all the player software performed equivalently. This is supported by every measurement method used. Subjectively with headphones attached to the DAC, I did not notice a difference listening to the music being played back while doing the DMAC Test.

2. No evidence of anomalies in the Dunn jitter test signal. This is not surprising as I had already previously reported that I was unable to detect more jitter with increased processor load as some seem to believe. From what I can tell, jitter is primarily a hardware property and software timing issues lead to obvious audio drop-out rather than subtle pico- or nano-second changes in the audio output.

3. Although I did not do an equivalent DMAC (DiffMaker) test with the DSD audio, it looks like all 3 programs tested with DoP capability performed equivalently using the RightMark test. Still waiting for more DSD content for this to matter. :-)

4. I see no evidence that special features like memory playback, "direct mode", "integer mode", "SysOptimizer" made any difference compared to the output from the no-frills TEAC player where I did not even turn on the memory playback feature with the 2008 MacBook Pro.

Bottom line is that these programs work well to output bit-perfect audio. The MAIN feature over iTunes is the ability to automatically adjust the sample rate. Beyond that, I'm happy to own both Decibel for its simplicity and flexibility in playing all kinds of formats as well as Audirvana Plus for the full feature set including DSD playback and DST decoding. I just don't see any evidence that they sound any different...

Do bit-perfect Mac audio players sound the same? Yes, as far as I can measure and have personally experienced.

Again, let me know if you have any evidence otherwise.

I was E-mailed shortly after publishing the TEAC UD-501 review if I've tried JPLAY - not yet, but in the weeks ahead may find some time to hook up the Windows setup and have a look...  Until then, I recommend reading Mitch's excellent writeup between JPLAY and JRiver.

Enjoy the tunes :-).


Addendum - June 9, 2013:
To answer that question of why even just -1 click at 8kHz with the iTunes equalizer resulted in such a low DMAC correlation null... Here's the answer:
Yes, you can see a very small dip in the frequency response at 8kHz - the intent of the EQ. However, the high frequency gets rolled off very significantly as well from ~15kHz.
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MEASUREMENTS: Digital Filters and Impulse Response... (TEAC UD-501)

June 5, 2013, 9:09 am
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By now, I suspect that most of us have read reviews and comments about the different types of digital filters available in many of the modern DAC's these days. A number of years ago - 2006 to be precise - I remember reading in Stereophile about these filter types and first saw the impulse response graphs in this article by Keith Howard. At that time, I remember being perplexed by all the variation of filter types possible but remembered how "cool" it seemed that here is another factor beyond jitter which may differentiate the "digital sound" from analogue... Furthermore, those graphs looked a little scary - Wow! Look at all that ringing!

Skip forward a few years and we see the introduction of new DAC's with various filter options available. I remember taking notice with the introduction of the Meridian 808.2 around 2009 with their "apodizing", minimal phase filter. In time, folks on forums would talk about how detrimental the whole "pre-ringing" would be with various manufacturers following the minimum phase filter design as selling points. Of course, in life, almost nothing is that simple; as if simply getting rid of one "issue" (like pre-ringing) would lead to joy and happiness without some price to pay. In terms of filter types, the price to pay includes a combination of early frequency roll-off and potential aliasing products.

As I have shown previously, the TEAC UD-501 has interesting properties including 2 digital filters based on the Burr-Brown/TI PCM1795 (SHARP and SLOW), as well as an OFF mode which results in removal of the oversampling filter and hence a "NOS" mode of operation. Today, let us have a look at the impulse response graphs, and consider issues like aliasing...  I even found another little surprise from this DAC!

First consider the sine wave frequency roll-off I previously found - this is why NOS DAC's sound a bit rolled off on the top end:


I. SHARP (Linear Phase) filter.

Let us start with the SHARP filter because this is the standard filter implemented in most DACs where you do not have a choice of settings. As you can see from the frequency graph above, this looks like the "classic" brick wall, and as you have no doubt seen, the impulse graph looks "classic" as well:

I played back a 16/44 "impulse" file and "recorded" it with my E-MU 0404USB at 24/192 to obtain that image above from Adobe Audition if you're wondering. Notice the positive deviation (up) which correlates with the phase of the impulse file therefore the TEAC maintained absolute phase. This is the standard "linear phase" filter with moderate amounts of pre and post-ringing. Here are graphs using white-noise to demonstrate the steep filter roll-off around the Nyquist frequency.

16/44 - White noise:

24/96 - White noise:

And here is what a 19kHz -10dB sine wave looks like with this filter measured up to ~100kHz (similar to John Atkinson's recent measurement changes in Stereophile which he discussed in the April 2013 issue, page 180):

We see a number of harmonics present with the largest being up at 38kHz (-90dB amplitude so obviously inaudible) representing one octave above the primary signal at 19kHz.

Although there is pre-ringing in the impulse response - hence lower temporal resolution, the good thing about the linear phase filter is that audible phase distortions are minimized. Also linear filters tend to measure very well compared to other implementations mainly because of the high frequency resolution which is what we usually look at with our FFT measures like frequency response, intermodulation distortion measurements, etc.

II. SLOW (Linear Phase) filter.

Next, let us consider the SLOW filter with a more gradual and earlier top end roll-off. I believe in some circles, the term "apodizing" may be used to refer to these filters although other places seem to also associate this term with minimal phase filters - if there's a simple consensus definition of "apodizing" as it refers to digital audio, please help clarify this for me. Here's the impulse response:

Nice...  It's still a symmetrical profile so it's a linear phase filter with low phase distortion. The (supposedly bad) pre-ringing is significantly reduced, hence better temporal characteristics, but here's the price one pays:

16/44 - White noise:

24/96 - White noise:

Notice that in both graphs, the roll-off doesn't hit the noise floor until well beyond the Nyquist frequency; we're looking at close to 35kHz for 16/44 and almost 75kHz with 24/96! No doubt this will result in aliasing distortions (the 19kHz -10dB graph):

There you go. Notice that high amplitude 25kHz aliasing distortion showing up now almost up to the level of the 19kHz primary.

So, basically we see a compromise with the SLOW filter...  Allow more aliasing, but also significantly decreasing the duration of the pre-ringing in the impulse response plot.

III. Digital filter OFF ("NOS").

The last of the TEAC UD-501 standard settings is of course with the digital filter OFF - the "NOS mode".

Impulse response:

Wow... No pre-ringing. It's essentially a square wave representing the duration of a single 16/44 sample. But in order to achieve this of course, the price to pay is tremendous:

16/44 - White noise:

24/96 - White noise:

Without any filtering, all the aliasing distortions get through. NOS DACs are said to sound "dirty" for this reason although of course many folks subjectively find the sound pleasing. Behold the 19kHz -10dB signal and all the aliasing and harmonics that gets through:

"Impressive"... Again, the price to pay for essentially having "perfect" impulse response with zero pre- or post-ringing. I would say that these effects are the most audible. So for those folks who really advocate for the "NOS sound", I guess it just means they prefer noisy, inaccurate reproduction with numerous harmonics and aliasing distortions. (I would love to see how these graphs look like on those expensive Audio Note NOS DAC's!)

IV. Minimal Phase filter!

Surprise! The TEAC UD-501 also has a minimal phase filter mode. This one is implemented by the 24/192 upsampling setting coupled with the digital filter turned OFF. Here's the impulse response:

As you can see, with a minimal phase filter, pre-ringing is not an issue. However, all that "energy" has been shifted into post-ringing. In fact, given the long ripple trail (about 4ms compared to SHARP filter 0.8ms start to finish of impulse), the absolute temporal resolution is the worst with this filter. However, since many believe that post-ringing is more "natural" and furthermore masked by the primary signal, its presence isn't particularly problematic (according to them). Here's the rest of the graphs:

16/44 - White noise:

24/96 - White noise:

Using an onboard ASRC chip, the upsampling algorithm does a excellent job achieving a sharp frequency roll off to prevent signals going beyond Nyquist.

Here's the 19kHz -10dB signal:

Very nice - even cleaner than the SHARP filter (as one would predict given the lower temporal resolution). Some harmonics present but no aliasing distortion.

What's the price to pay for minimal phase filters? The post-ringing duration is quite long in this implementation. Also, given the non-linear characteristic, there may be audible phase distortion.

V. So what?

Yup... Now you know what the graphs look like and the options available on the TEAC DAC (and of course variants of these filters with other DACs)...

So what?

Objectively, I think it can be said that the fact that NOS DAC's don't just plain sound awful really speaks to just how forgiving our ears are!

Subjectively, as one who aims for accuracy in waveform reproduction, I've found that my "favourite" remains the standard linear phase (SHARP) filter both based on what I hear and intellectual satisfaction. I've never been a fan of the NOS sound - the rolled off highs and slight "dirtiness" to the sound quality robs it of resolution IMO (I presume this is secondary to the high aliasing distortion). I will however give a thumbs up to the TEAC engineers for the minimum phase upsampling with digital filter turned OFF. Some audiophiles have commented that in many DAC chips, it's actually not the pre-ringing that is an issue, but rather the processing of the digital filters themselves and taking out these built-in filters improves the sound quality... At least there's that opportunity on the TEAC - I'll have to listen a bit more...

One final thing. Consider this, just how audible is pre-ringing anyways? Realize that music isn't an impulse so these graphs are artificial exacerbations of the "ringing", just like testing with square waves generally do not show perfect instantaneous transitions in real life equipment. But even if I zoom into that SHARP impulse response (note I purposely inverted this impulse), this is what I see:


We have 7 waves over 60 samples, measured at 192kHz. This means we're looking at low amplitude pre-ringing at around the Nyquist frequency ~22kHz (remember 16/44 impulse signal). Hmmm, what kind of human physiology has the capability to hear that lead up to the primary impulse wave? Has anyone ever proven this is audible in music? Any references to controlled trials? Even the golden ears at Stereophile seemed unclear about the significance of these filters in that 2006 article before folks started claiming pre-ringing was "bad" (they were even listening to maximal phase filters with lots of pre-ringing and didn't dislike that nor show clear preference to the minimal phase filter).

As usual, if anyone has information/data/results about the audibility of these digital filters, please drop a note!

Time to go listen to  those 2013 Eagles remasters now...

(PS: Looking at the oversampling function of this TEAC DAC, I was hoping ASUS could have done something similar with the XONAR Essence One as I noted a few months ago - instead they made a gimp upsampler which rolled off way too early! Unfortunate.)

Addendum: As suggested by Shoddy in the comments - if I just have a look at the pre-ringing waveform from the SHARP (linear phase) filter, boost the levels and check the FFT; here's what it looks like:

Indeed, most of the energy of the pre-ringing waveform seems to be clustering around the Nyquist frequency and measures about 10 dB louder than what I assume is low frequency noise...
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Getting There... (Early HT Room Setup)

November 20, 2013, 10:11 pm
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Back from my overseas business trip late last week. It's going to be really busy since I'll be moving to the new house in 1 week. Massive amounts to pack up!

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




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

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

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

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

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

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

May 22, 2013, 8:40 am
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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...


--------------------------------
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:

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Search

MUSINGS: A Look At The Sound Room...

December 13, 2013, 10:05 pm
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It's about 2 weeks now since I moved into the new home. As expected, it was stressful; with kids in tow, it's not like the bachelor days with just one's personal belonging in the trunk of my old hatchback!

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

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

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

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

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

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

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

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

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

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

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

II. Multichannel signal path:

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

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

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

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

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


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

III. Challenges...

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

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

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

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


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

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

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

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

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

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(BTW: I got my Nikon D800 back from Nikon Canada for repairs on an autofocus issue under warranty. Wow. The focus seems to be spot on and only minimal lens fine-tuning is required now. There was quite a stir online about poor left auto-focus point accuracy as well which seems to be much better now. If you have a D800 and are running into focus issues, check if Nikon can do a tune-up.)
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MUSINGS: Those "next generation" game machines - PS4, XBOX One, Wii U...

December 25, 2013, 1:27 am
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Battlefield 4 - PC gaming time in the living room... Silverstone LC14 case in lower shelf. Arcade Street Fighter stick to the left. Old SNES still in the box to the right :-)
As I opened boxes and put things into place after the recent house move, I pondered about the living room situation.

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

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

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

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

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

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

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

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

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

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

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

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

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

[Update Dec. 29, 2013]: I just watched the documentary Indie Game: The Movie. Wonderful snapshot into the world of independent gaming! I really enjoyed the quirky Super Meat Boy a couple years ago :-).
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MEASUREMENTS: USB Cable Extension with Ethernet Cables - does it worsen jitter?

December 29, 2013, 11:50 am
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I mentioned in my "look at the sound room" post the other day that in the Emotiva XSP-1's multichannel "HT Bypass" mode, I'm dealing with some noise when the USB DAC (TEAC UD-501) is connected. In an attempt to fix the problem, I've tried a number of ways to minimize the problem - different USB ports on the motherboard, USB3 vs. USB2 ports, different USB cables, ferrite cores attached to the USB cables, even put in a USB3 PCI-E card... I've even dissected an old USB cable to disconnect the 5V line but it looks like the TEAC needs this present. All to no avail - the high pitched whine and buzz continued.

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

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

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

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

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

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

Setup

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

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

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

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

 

Results

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

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

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

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

No difference!

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



Yet again - essentially no difference.

Summary

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

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

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

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

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

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

Enjoy the tunes.

[2013/12/31 Update]

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

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

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


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

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

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

January 12, 2014, 9:12 am
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PBK-1 mic about to record some subwoofer beeps and boops...

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

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

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

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

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

I. Paradigm PBK-1 (Perfect Bass Kit)

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

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

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


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

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

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

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

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

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


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

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

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

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

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

After REW EQ:

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

Looking Ahead...

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

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

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

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

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

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

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

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

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

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

January 12, 2014, 5:13 pm
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Mobile space is where it's at these days for major consumer computing innovations... New advancements in wireless communication, low power, more speed. Amazing just how fast progress is being made! It was only in early 2000 that the 1GHz barrier was broken for consumer desktop CPU's, 2005 was the first mainstream dual-core x86 CPU (both AMD X2 and Pentium Extreme 840). Now, we've got >1GHz quad-cores with 1080P screens on handheld devices in less than 10 years... (Yes, I know an ARM CPU isn't directly comparable to the x86, but still impressive nonetheless!)

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

I. The 'Victims'

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

 

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

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

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

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

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

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

II. Results

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

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

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

RightMark summary:

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

16/44:


24/96:

Jitter:


III. Summary

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

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

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

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

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

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

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

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

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