Archimago's Musings

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.

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...

In the previous post, Frans de Gruijter posed an interesting comment and question...

"When for instance a sine wave is used of say 5kHz which is stopped at the 0 crossing what would the ringing be?"

Interesting how I've never seen measurements of this... We always see just the worst case scenario with a single maximum amplitude impulse response when as I mentioned previously, this kind of thing does not exist in real music.

So then folks, let us have a look at what a single "pulse" looks like with the different filters at 5kHz and 10kHz. Again, remember that I am measuring the waveform back using the E-MU 0404USB at 24/192 so there will be some "ringing" imposed by the ADC itself when looking at the TEAC UD-501 NOS mode especially.

Note: Ignore the phase inversion between 5kHz and 10kHz - I accidentally caused this inversion and it's not due to the TEAC hardware.

I. SHARP (Linear Phase) Filter:

Reminder of the impulse measurement:

5kHz (16/44):

10kHz (16/44):

Ringing evident at 10kHz but lower amplitude and shorter duration than impulse (ie. worst case scenario). Notice that the frequency of the ringing is the same as the impulse so we're looking at around Nyquist (22kHz).

II. SLOW (Linear Phase) Filter:

Impulse response:

5kHz (16/44):

10kHz (16/44):

III. Digital filter OFF ("NOS-mode"):

Impulse response:

5kHz (16/44):

10kHz (16/44):

IV. Minimal Phase Upsampling to 24/192 with Digital Filter OFF:

Impulse response:

5kHz (16/44):

10kHz (16/44):

V. Summary...

Some comments about the linear phase pre-ringing which some folks obsess over:

1. Lower frequencies like 5kHz essentially is associated with no pre-ringing to worry about in linear phase filters... Since the ear is most sensitive to tones from 1-5kHz, this may be reassuring. As you can see, pre-ringing does show up higher like at 10kHz.

2. Ringing amplitude also correlates with the frequency. The lower the frequency, the less the pre-ringing amplitude. For example, the SHARP filter 10kHz pulse pre-ringing amplitude is ~1/2 of the impulse.

3. The pre-ringing frequency itself remains high at around Nyquist like the impulse response so I remain skeptical that it's even audible (seriously folks, real musical recordings have a noise floor as well as complex harmonics - how anyone can claim that it's low-amplitude, high frequency pre-ringing causing any defect in the sound is a mystery to me!).

Of course, there is an "easy" way to not have to worry about the pre-ringing phenomenon in the audible spectrum... Go download a hi-res copy of the music ;-).

SHARP filter, 10kHz sampled at 88kHz:

Presto! Pre-ringing gonzo!

Note: Actually, there will still be some very high frequency pre-ringing for 20kHz sampled at 88kHz but we're talking 40+kHz ringing at even lower amplitude... I'm pretty sure I can live with this! (Again, please ignore the phase inversion compared to the graph above.)

Close to a month back, after publishing my TEAC UD-501 results, someone E-mailed me about the use of JPLAY with this DAC.

Although I had not tried the program yet, it begs the question, how's it possible that JPLAY could sound any different? JPLAY is described as a bit-perfect player openly discussed in the FAQ but if you look at items 5 & 6, they seem to imply that "timing is everything" and somehow, an optimized software player can make a difference. Note that already there have been tests questioning these claims (here's a nice one from Mitch over on the Computer Audiophile between JPLAY vs. JRiver). The fact is, there is so much going on the in the hardware level such as OS and USB buffering, DAC buffering, driver-interface interactions that are just beyond the "reach" of the player software that it really makes no sense IMO for any software developer to make such claims!

Since my hardware was already set-up with the Mac, I did the software measurements with that platform first (results here). This week, I turned my attention over to the Windows platform. Here then are the candidates for the test:

1. foobar2000 - Free, highly configurable, fully functional, a myriad of plug-ins including support for all major file formats, this is my daily workhorse for playback on my Windows machines. I generally use the ASIO plugin for most of my DAC's to ensure bit-perfect data transfer. The dynamic range (DR) plug-in comes in very useful to check for severity of dynamic range compression among different "pressings". ABX comparator is great for controlled evaluation of sound quality. In terms of sound "quality", foobar intends to "just work". It makes no pretensions about sounding "better" than anything else - just bit-perfect - nothing more, nothing less. For this test, I ran version 1.2.6 with ASIO plugin version 2.1.2. All settings set to default. Tested with the ASIO driver and WASAPI (event style). Furthermore, I installed the SACD Decoder plug-in for DSD playback - works great (remember to run the ASIOProxy plugin, configure the ASIO settings, and go to Tools --> SACD to make sure it's not transcoding to PCM).

2. JRiver Media Center for Windows - I previously looked at this player for Mac. This is of course the "original" version since the PC release came first and only recently ported over to OS X. I bought this program since it allows me to easily play DSD64 and DSD128 to the TEAC DAC with DoP including DST decoding. I'll be testing version 18.0.177. Tested with ASIO, Kernel Streaming, and DSD playback.

3. iTunes x64 for Windows - Why not :-). Using version 11.0.3.42. It bears repeating - the lack of native FLAC makes iTunes essentially useless for me since I refuse to use ALAC (which doesn't compress as well by default) and there's no way I'm going to leave lossless files uncompressed as WAV or AIFF - ridiculous waste of HD space. Just like with the OS X tests, I've converted the test audio to play as AIFF files, volume at 100%, all plug-ins turned off. To handicap iTunes even further, I'm going to play these AIFF files for measurement off a comparatively slow Patriot Rage XT USB stick rather than copying them over to the SSD. To potentially make it even worse, I tested with DirectSound - manual setting of 16/44kHz and 24/96kHz (royal pain folks - you have to adjust the default Windows settings, iTunes settings, and make sure to restart iTunes each time).

4. TEAC HR Audio Player for Windows V1.0 - this is the TEAC freebie for use with their DAC's. Will also play DSD using either the TEAC's native ASIO driver or DoP. This program does not support DST compression for DSD unfortunately. Like with the OS X version, memory playback turned OFF.

5. JPLAY v5.1 trial - self-described as the "hi-end audio player for Windows" - implicit in this is the idea that somehow this program is capable of better sound quality. As I quoted earlier, the web site suggests that "timing is everything", which implies that today's multi-gigahertz multi-core computers somehow have issues dealing with relatively slow data processing involved in even 24/192 bitrates. Various audiophile sites have obliged with reviews claiming that this program makes digital audio "almost neck and neck with vinyl" (sure, vinyl sounds great with a good system, but is that the definitive standard?!). We shall see just whether any difference can be detected using this program in each of its River, Beach, Xtream, and ULTRAstream "engines". I used foobar2000 as the front end since JPLAY can be used with any program that's compatible with ASIO. I'll even try with full optimizations like hibernate mode, high priority, and low buffer sizes! Thankfully, the trial version only interrupts playback every 2 minutes which should be enough for the test "runs".

Because JPLAY brings with it so many options, I am actually going to split this topic into 2 sections - this first part is getting too unwieldy... Watch for Part II in this series focused on JPLAY.

Setup:

(Similar to previous DMAC Test and OS X Player Test.)

ASUS Taichi (*running audio player*) --> shielded USB (Belkin Gold) --> TEAC UD-501 DAC --> shielded 6' RCA --> E-MU 0404USB --> shielded USB --> Win8 Acer laptop

The ASUS Taichi DH51 is the same machine I used in the laptop tests. CPU is the Intel i5-3317U (1.7-2.6GHz dual core, 3M cache). "Only" 4GB DDR3 RAM (good enough for audio memory play IMO). Of course, the machine will not be multitasking with other programs running during playback apart from the usual OS tasks. OS is Windows 8 x64 with all recommended updates as of June 1, 2013. Tests were done with the laptop unplugged running off batteries.

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.

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

Like with the Mac tests, all the test audio was encoded with FLAC except for the iTunes test presented as uncompressed AIFF.

PCM 16/44 Summary:

As you see, the leftmost item is from the Mac test with Decibel, the rest of the recordings done with the Windows 8 platform... Look at how close the results are despite measurements taken about 3 weeks apart. JRiver "KS" refers to Kernel Streaming.

Frequency Response:

Noise:

THD:

Stereo crosstalk:

PCM 24/96 Summary:

Again, the leftmost item is done with Decibel on the Mac. The rest are all from the Windows 8 machine.

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

DSD64 via either ASIO native or DoP for the programs that support DSD:

This is done 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 or natively with ASIO and measured with RightMark. As usual, we see the effect of noise shaping in DSD up in the ultrasonic range.

Yet again, leftmost column is from the Mac using JRiver (OS X) playing DSD via DoP (on the Mac, DoP was the only supported way to play DSD). The foobar SACD/DSD plug-in worked very well for me, as did JRiver (Windows) and of course TEAC's own player software.

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

DSD output looks good across the board.

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

foobar ASIO:

foobar WASAPI:

JRiver ASIO:

JRiver Kernel Streaming:

iTunes (with uncompressed AIFF rather than FLAC):

TEAC HR Audio Player:

I don't see any differences in these spectra. Certainly no major sidebands creeping up. Although not shown, the 24-bit spectra look unremarkable also.

Part III: DMAC Protocol

As the reference, all comparisons were made to a recording with foobar ASIO.

Very high quality correlated null depth in all the players in the mid-80+dB range in this "machine listening" test. MP3 320kbps was used in the last column as a comparator - measuring in the mid-60dB level as usual.

Part IV: Conclusion

Ultimately I cannot say much beyond what was expressed in the OS X conclusions. With these Windows programs, the players are all capable of indistinguishable high quality audio output to my TEAC UD-501 whether 16/44 and 24/96 PCM or DSD using DoP or native ASIO. Furthermore, in the Windows world with the various driver models, I detected no significant difference between ASIO, WASAPI, or Kernel Streaming whether through FFT-based RightMark analysis, "microscopic" examination with the Dunn J-Test, or "macroscopic" listening test over >30 seconds with the DMAC.

Although DirectSound does not claim to be "bit-perfect" since it takes the integer audio --> converts to 32-bit float --> dithered back to 16/24-bit and sent to the DAC through Windows Mixer at a specific sample rate (set in your Control Panel for the sound device), it looks like it is able to do this with a single audio stream of the same sample rate without significant deterioration in the output in the 24-bit domain (remember the DMAC Test is 24-bit audio) - not exactly rocket science so this is to be expected. I believe many folks feel the quality of the Windows Mixer has improved over the years. Dithering down to the 16-bit domain would likely be very detectable in the measurements and would have messed up the 16-bit J-Test as well - this is why I always keep my Windows default output as 24-bits (note that the TEAC driver does not have a 16-bit setting for DirectSound so I can't demonstrate what 16-bit dithering looks like). Of course if you have multiple streams going through the mixer, things could deteriorate - but this is not generally relevant for home music playback. Also, if you run a DTS or AC3 file through DirectSound, it would not be surprising to hear errors in the bitstream.

Over the years I have used foobar, JRiver, and even iTunes for hours of listening... Other than the tests here, I have never tried to perform any controlled testing. However, I have not found occasion to complain that the audio output sound "bad" comparatively. Personally, I like using ASIO since there's just less risk of messing up the settings.

The simple message remains - stay bit-perfect and stop worrying. So far the most important factors I have seen with digital gear has been getting a good DAC with good drivers and make sure there are no settings lingering around that could be messing up the sound. Not only is there no objective difference between Windows audio players so far, but this is also shown to be cross platform with the Mac. No need for flame wars between Win and Mac; they're the same.

As usual, I invite anyone to comment if they think these results and/or conclusions are erroneous based on controlled testing results.

Lets get "extreme" next time and consider JPLAY - could there be a measurable difference?

My music selection tonight: Hilary Hahn & Oslo Philharmonic's Mendelssohn & Shostakovich Violin Concertos SACD (2002 - sadly it looks like the 2.0 layer came from PCM 44kHz source!).

Remember to enjoy the music folks...

There's something happening here,What it is ain't exactly clear.-- Buffalo Springfield

Welcome to Part II of the "Bit-Perfect" roundup for Windows.

In this installment, I'll focus on JPLAY (most recent version 5.1) which has been highly promoted around many of the audiophile web sites like 6moons, EnjoyTheMusic ("First Step to Heaven = JPLAY"!), High Fidelity (Polish), TNT-Audio, recently AudioStream plus all the hoopla around the terribly misleading series of articles on computer audio by Zeilig & Clawson in "The Absolute Sound" in early 2012.

If you browse around the JPLAY website (or hang out on various computer audiophile forums), you run into suggestions that process priority, memory playback, task switching latency, driver type (eg. ASIO, WASAPI, Kernel Streaming) all have some kind of effect on the playback audio quality (usual digital whipping boy for bad sound quality is of course the dreaded jitter). Sure, computers can be complicated, but is it possible for these factors to affect audio output significantly enough to hear and common enough to worry about with a modern USB DAC like in my test set-up? Maybe, but like many subtle things in life "reported" to be true, it's hard to know with confidence unless systematically tested. To address these "issues" JPLAY even has a mode where the screen gets turned off, OS processes are halted, drivers turned off, etc. the famed "hibernate mode". Since the keyboard is turned off, the computer comes back to life after the music has completed playing or if you remove a USB stick during playback! Realize that this makes the computer even less interactive than a disc spinner!

Mitchco has already used the DiffMaker program to compare JPLAY and JRiver about a year ago but was not able to detect a difference. Likewise, there have been some heated discussions over on Hydrogen Audio regarding this program.

Well, let us put this program and various of its settings on the test bench and see what comes up...

Firstly, the setup - same as Part I:
ASUS Taichi (*running JPLAY/foobar*) Win8 x64 --> shielded USB (Belkin Gold) --> TEAC UD-501 DAC --> shielded 6' RCA --> E-MU 0404USB --> shielded USB --> Win8 Acer laptop

Remember that JPLAY is essentially a "playback engine" using its own buffering algorithm and runs as an Audio Service under Windows. In this regard, it is quite unique. By itself, it has a very bare-bones text-based interface. Therefore, for ease of testing, I'm going to be using JPLAY through foobar2000 as an ASIO output "device". Within JPLAY settings, one can then specify the actual audio device and which driver to use (eg. ASIO, WASAPI, Kernel Streaming...), along with specific parameters like buffer size. There are four "engines" - Beach, River, Xtream (Kernel Streaming only), and ULTRAstream (Kernel Streaming & Windows 8 only) - I am unaware of how they are supposed to differ. Note that for all these tests, I had "Throttle" turned ON which is supposed to increase priority to JPLAY (and hence diminish task priority of other processes). Volume control was turned OFF. According to the manual, there are even "advanced settings" called DriverBuffer, UltraSize, and XtreamSize which I did not bother playing with since I presume they're deemed less important if one has to go into the Registry to adjust. Thankfully, the trial version will play ~2 minutes of uninterrupted audio before inserting ~5 seconds of silence. This is enough time to measure the audio quality using my standard tests.

I. RightMark 6.2.5 (PCM 16/44 & 24/96)

Let us start with JPLAY using the ASIO & WASAPI drivers. I think it is important to remember that the makers of JPLAY recommend using Kernel Streaming instead. My suspicion is that JPLAY essentially just passes information off to these ASIO and WASAPI drivers so logically there would be little reason to believe measurements & sound quality should change at all. Furthermore, the CPU utilization for the process "jplay.exe" with ASIO and WASAPI remained low during playback - on the order of <2% peak and usually <1% using the i5 processor. As we see later, this dramatically changes with Kernel Streaming where the program can actually get closer to the hardware and do direct processing in "kernel mode".

Here's the data for JPLAY at 16/44 with ASIO and the two "engines" that support ASIO - "Beach" and "River". When you see a number like "ASIO 2", that refers to the buffer setting (number of samples). The software recommends using the smallest buffer size with "DirectLink" being 1 sample. I was not able to get the DirectLink setting working without occasional blips and errors once I started doing Kernel Streaming, but 2 works well for 16/44, 4 was good for 24/96 so I standardized on those settings throughout. As you can see, I also measured with buffer of 64 samples to see if that made any difference (I see from the manual that Xtreme only works for buffer sizes up to 32 samples - it didn't complain when I set it to 64):

Note that I used the standard foobar2000 + ASIO audio output as the comparison measurement in the leftmost column.

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

For some reason, WASAPI refused to initiate for me at 16/44. But it did work at 24/96, so I have included that here as well. Again, foobar + ASIO was used for comparison on the left:

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

So far, no real difference between the JPLAY playback and foobar2000.

Now, lets deal with Kernel Streaming. It is with KS that we see significant CPU utilization by "jplay.exe". Instead of <2% peak CPU utilization above, with low buffers like a setting of 2 samples for 16/44, I see peak CPU utilization of 13%, and with a 4-sample buffer at 24/96, peak CPU jumps to 16% with my laptop's i5 and the "Xtream" engine. The "ULTRAstream" engine chews up even more CPU cycles by another 1-2% above those previous numbers. The moment the buffer size was increased to 64 samples, CPU utilization dropped to 3% with 16/44 and 6% with 24/96 peak. It looks like only when Kernel Streaming is used, does JPLAY actually kick in to maintain those tiny buffer settings.

Starting with 16/44, here's Kernel Streaming with mainly the "Xtream" and "ULTRAstream" engines since these two cannot be used with ASIO. I included "Beach" in these measurements as well out of interest. The test on the far right with "hiber" refers to the use of the "hibernate mode" where the computer screen, OS processes, drives, etc. turn off during playback. Again, foobar + ASIO was used as the comparison:

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

Here is 24/96:

Frequency Response:

Noise:

THD:

Stereo Crosstalk:

So far, despite all the changes in CPU utilization, different audio "engines" and buffer settings, I see no substantial change in the measurements that would suggest a qualitative difference in terms of the analogue output signal from my DAC.

JPLAY supposedly can support DSD playback. I did not test this function.

Part II: Dunn J-Test for jitter

I did a whole suite of J-Test with all the different audio "engines", either 2 or 64 sample buffers, ASIO/WASAPI/KS, also tried the "hibernate" mode. For brevity, here's a selection of 16-bit jitter spectra using various settings:

foobar2000 ASIO:

Beach ASIO 64-sample buffer:

River ASIO 2-sample buffer:

Xtream Kernel Streaming 64-sample buffer:

ULTRAstream Kernel Streaming 2-sample buffer:

ULTRAstream Kernel Streaming + Hibernate 2-sample buffer:

Essentially no difference in the J-Test spectra.

Recall that the 24-bit Dunn J-Test is done with a 24/48 signal. While doing this test using Kernel Streaming mode, something strange was found.

This is what the 24-bit Dunn J-Test looks like with foobar2000 + ASIO (notice primary signal at 12kHz rather than 11kHz with the 16-bit test):

The 24-bit LSB modulation signal is buried under the noise level. This is quite normal.

Here is Beach + ASIO:

Again, quite normal - we're still using the TEAC ASIO driver.

Look what happens with 24-bit J-Test Beach + Kernel Streaming (doesn't matter what buffer size):

Eh? What's with all the modulation signal bursting through like with the 16-bit test?!

My suspicion is that JPLAY isn't handling the last 8-bits in the 24-bit data properly... One possible scenario is where the last 8 bits got flipped from LSB to MSB, thus causing the LSB signal to show through at a higher level. With RightMark, this is what it looks like:

We've "lost" the last 7 bits of resolution at 24/48 with Kernel Streaming causing the dynamic range to drop to 102dB (17-bits resolution) instead of the usual >110dB (24-bits into the noise floor). I considered the possibility that this may represent some sort of dithering but why would it be applied to 24/48 and not 24/96?

Strangely, this anomaly did not show up at 24/96. I did not bother to check if 88/176/192kHz sampling rates were affected.

Part III: DMAC Protocol

Time for the machine listening test of 24/44 composite music using Audio DiffMaker. As shown in previous blog posts, this test seems to be quite good in detecting even relatively small changes like minute EQ adjustments, difference between AAC, MP3, etc... This test is also similar to what mitchco did last year.

Here are measurements of a few settings in JPLAY in comparison to foobar ASIO as reference. I included the most "extreme" one that I could perform - ULTRAstream + 2-sample buffer + hibernate mode. As usual a couple of "sensitivity tests" with MP3 320kbps and slight EQ change in foobar for comparison:

The machine was able to correlate the null depth down to around 90dB with all the JPLAY settings and foobar suggesting no significant difference in sound in comparison to the reference foobar + ASIO playback.

Part IV: Conclusion

Based on these measurements, a few observations:

1. 24/48 with Kernel Streaming appears buggy as shown with the 24-bit J-Test and RightMark result. Don't use it (as of version 5.1) if you want accurate sound. However, subjectively it still sounds OK to me since it's still accurate down to 16-bits at least. ASIO works fine. 24/96 is fine. I don't know if other sampling rates with 24-bit data are affected. For some reason I could not get WASAPI 16/44 to work with JPLAY even though it was fine with foobar2000.

2. Technically, JPLAY appears to be bit perfect with 16/44 and likely 24/96 based on my tests (of course we cannot say this for 24/48). Since the program claims to be bit perfect, this is good I guess.

3. I was unable to detect any evidence of sonic difference at 16/44 and 24/96 compared to a standard foobar set-up. RightMark tests look essentially the same. Over the months of testing, I see no evidence still that software changes the jitter severity with CPU load, different software, even different DACs (as I had postulated awhile back in this post). DiffMaker null tests were also unable to detect any significant difference in the "sound" of the analogue output from the TEAC UD-501.

4. Still no evidence that extreme settings like "hibernate mode" which reduces the utility of the computer makes any sonic difference. Of course, it's possible that this could make a difference with some very slow machines like a 1st generation single-core Atom processor with small buffer settings doing Kernel Streaming... But in that case, why not just increase the buffer size with Kernel Streaming (why all the "need" for low buffer settings and obsession over low latency for just playback?!) or just go with an efficient ASIO/WASAPI driver? I'd also recommend a processor upgrade if you're still rocking an old Atom!

Over the 3 nights I was performing these tests, I took time to listen to music with the various JPLAY settings - probably ~3 hours total. It sounds good subjectively (other than the brief interruptions every 2 minutes or so for the trial version). The i5 computer shows a little bit of lag with Kernel Streaming and low buffer size if I'm trying to do something else. I listened to Donald Fagen's The Nightfly and Peter Gabriel's So (2012 remaster) in 24/48 with Kernel Streaming and didn't think they sounded bad despite the issue I found (remember, the anomaly was down below 16-bits). Dire Strait's Brother In Arms sounded dynamic and detailed as usual. Likewise Richard Hickox & LSO's Orff: Carmina Burana (2008, 24/88 SACD rip) sounds just as ominous. (Almost) Instantaneous A-B'ing is possible in foobar changing ASIO output from the TEAC driver to JPLAY and I did not notice any significant subjective tonal, amplitude, or resolution difference using my Sennheiser HD800 headphones with the TEAC UD-501 DAC flipping back and forth a number of times.

Bottom line: With a reasonably standard set-up as described, using a current-generation (2013) asynchronous USB DAC, there appears to be no benefit with the use of JPLAY over any of the standard bit-perfect Windows players tested previously in terms of measured sonic output. Nor could I say that subjectively I heard a difference through the headphones. If anything, one is subjected to potential bugs like the 24/48 issue (I didn't run into any system instability thankfully), and the recommended Kernel Streaming mode utilizes significant CPU resources when buffer size is reduced (which the software recommends doing). I imagine that CPU utilization would be even higher if I could have activated the DirectLink (1-sample buffer) setting.

Finally, there's the fact that this program costs €99. A bit steep ain't it? JRiver costs US$50, Decibel on the Mac around $35, foobar2000 FREE and these all feature graphical user interfaces and playlists at least! Considering my findings, I'm unclear with what DAC or computer system one would find tangible benefits after spending €99 for this program.

As usual, I welcome feedback especially with objective data or controlled test results (any JPLAY software developers care to comment on how the audio engines were "tuned"?). I would also welcome independent testing to see if my findings can be verified on other hardware combinations (especially that 24/48 issue).

Music selection tonight: Paavo Järvi & Orchestre de Paris - Fauré: Messe de Requiem (2011). Lovely rendition of Pavane for mixed choir!

As usual... Enjoy the tunes... :-)

Back in April, I posted my USB cable tests (note this was updated recently with the TEAC DAC & Belkin Gold results). To recap, basically I could not find significant analogue output differences between a few cables despite differences in construction and lengths - including one of which consisting of 2 cable extenders and totaling 17 feet in length. The analogue output from the DAC did not show significant change in frequency response, distortion, noise levels, or jitter whether the USB cable was used to feed an asynchronous USB-to-SPDIF device (the CM6631A), directly to an older adaptive isochronous DAC (AUNE X1), or directly into a modern asynchronous USB device (TEAC UD-501). Since it is these analogue waveforms that get transduced to sound waves, it's not a stretch to conclude that USB cables make no audible difference. My subjective evaluation of USB cables is consistent with the measured results - no audible differences in controlled tests (my own blind tests).

These days, we all use USB cables of different lengths and varieties for high-speed devices like hard drives and generally either it works or it doesn't. USB protocol sends data in packets which consists of not just the audio data itself (up to 1024 bytes at a time for high speed mode), but CRC check to detect errors, flow control, and also address information to direct the data to the appropriate device (remember, you can attach up to 127 devices to each host controller). The low-level details of this communication including timing are addressed by the hardware and generally outside of the purview of the software we install on the computer. Logically this means that fine timing issues (like data flow control and scheduling of the data packet delivery) would be outside of the effect of things like audio player software. Furthermore, with modern DACs, the machine will use its own internal memory buffer and very fine timing (like pico- and nanosecond jitter) will be derived from the accuracy of this internal clock.

USB data transfer rate for audio is much less demanding than something like a hard drive. Whereas we can easily transfer >200Mbit/sec to the hard drive using a simple generic USB2 cable, sending standard 16/44 stereo audio to a DAC is only ~1.5Mb/s, 24/96 only needs ~4.6Mb/s, and 24/192 ~9.2Mb/s. If we go all out with the TEAC UD-501 DAC with DSD128 or 32/384, even that "only" takes up to ~25Mb/sec.

Not unexpectedly, some message forums got a bit heated when the USB cable test post was published since obviously the results deviated from expectations or experiences of subjective evaluation or maybe the person is trying to sell something related. I have seen no good evidence from controlled tests demonstrating otherwise.

Shortly after that post, this set of measurements came out from one of the principles of Empirical Audio (I thought he promised to measure the AudioQuest Diamond USB cable?). Nice to see he measured the down-to-earth Belkin Gold anyhow. I will let you, fine readers, examine the data and determine if you think this correlates with audio quality, or just demonstrating minute differences between lengths of copper with little correlation to digital data transfer much less the analogue output from a DAC! (Note that he's measuring 5m [~16.5 feet] lengths of USB cable to get those picosecond numbers.)

For this installment of MUSINGS, I wanted to have a look at a recent article from the July 2013 issue of Hi-Fi News where they did a "Group Test" looking at USB cables. Since I do not subscribe to this magazine, I want to thank "Mushroom_3" and "Julf" off the Slim Devices/Squeezebox forum for bringing up this article.

As you can see on the cover - "USB cable sound - Fact or fantasy? p43" was advertised. So from page 43-51 (with 2 pages of ads), they tested 10 cables.

Here's the first 2 paragraphs of the article (as usual, I reproduce only small portions out of fair use for the purpose of discussion, education, and critique):

That's a pretty gross generalization and assumption to start off with isn't it? The only thing I think "every seasoned audiophile" knows is that without cables, there can be no sound - and that's assuming your system isn't a boombox :-). That's about the extent of discussion as to whether USB cable sound could be "fantasy" (I think the better word would be 'inaudible'). From there on, it's all about oscilloscope measurements (they give data risetime measurements for the cables and use this and the eye pattern as their objective data), and minimal detail was provided as to how they conducted the "blind" subjective test - we do not know how many subjects participated, how they were selected, in what way was the 'blinding' accomplished, duration between cable swaps, etc. Furthermore, we do not know how the participants scored the cables or what the statistical outcome is - it may be just a "blind" solicitation of subjective opinions on those 10 cables (a huge number to test properly!).

I appreciate that at least Paul Miller the editor spent a page talking about the technical bits of USB cabling on page 98. At least the physical cabling characteristics were described in detail but strangely nothing on the packetized nature or how they would correlate audible differences in such a system of data transfer since IMO this would be much more educational.

The comments below will hopefully make sense to those who have not read the article.

1. From what they wrote, the test setup went like this:
Laptop --> Test USB Cable --> Musical Fidelity M1 S-DAC --> coaxial cable --> Devialet D-Premier --> speakers
Why was the decision made to use the Musical Fidelity M1 S-DAC as a USB-to-S/PDIF converter? Seriously, you're testing USB cables but introducing a coaxial S/PDIF interface into the middle of the signal path for no clear reason when the M1 is already a fine DAC?! We know that a good USB asynchronous interface has lower jitter in general than most S/PDIF, so why not use the analogue output of the M1 directly to a preamp/amp? I would not be surprised if the Musical Fidelity is a better DAC technically than the Devialet (see below). There was no mention of what coaxial cable was used - surely this must be important since this is the digital cable directly connected to the DAC and these guys believe S/PDIF cables make a difference as per the first paragraph quoted above. How is it possible to test USB digital cables if there's potentially an even more jittery (yes, the dreaded jitter bogeyman yet again) digital coaxial cable/interface in the audio chain?

2. Why was the Devialet D-Premier used? I suspect why they used the USB-to-S/PDIF interface is because the Devialet internally resamples analogue to digital as per the discussion in the Stereophile review, plus it doesn't have a USB interface. Furthermore, the measurements for the D-Premier isn't all that remarkable as a DAC. It achieves a respectable 18-bit dynamic range (remember, the SB Touch already can do 17.5 bits) but like John Atkinson say - it's "not quite up to the standard set by the best-measuring standalone processors, such as the Bricasti M1, MSB Diamond DAC IV, NAD M51, or Weiss DAC202". I don't know about the M1 S-DAC specifically, but the Musical Fidelity M1 appears to outperform the D-Premier already (Stereophile data) and "offers performance that is close to the state of the art."

3. Why was the "grotty giveaway USB cable" measured on p.98 not part of the blind listening test? I wish they included a picture of this "grotty giveaway" - what makes it "grotty" anyways? Why not also tell us what that cable's data risetime is while you're at it? Doesn't this often seem like the case with these audiophile "shootouts"? They have all these high priced options (well, at least they included a £18 QED Performance Graphite) but neglect to include the real competition - something that is essentially free and works! Another idea - how about testing a reasonable quality "non-audiophile" USB cable like the Belkin Gold for <£5?

4. USB data risetime & eye pattern: is it even relevant to the analogue waveform coming out of the DAC? Are they measuring something which actually COULD have sonic impact? In terms of this article with the risetime derived from the eye pattern as the prime objective measure, what evidence do we have that this actually correlates with the subjective impression? So what if the £18 QED scores 12.8ns, £60 Kimber B BUS Cu scored 12.4ns, £139 Audioquest Carbon scored 11.9ns (hey folks where's the Diamond?), and the insanely priced £6,500 Crystal Absolute Dream scored 11.0ns? What does this risetime have to do with the final DAC analogue output anyways given the nature of packet digital transmission, asynchronous protocol, and a cable that can provide much more bandwidth than required for audio - especially since you're passing the bits off through a digital S/PDIF coaxial in the audio chain?! I question if taking "a good few months" (p.98) to develop this test was time well spent!

5. Why does there appear to be so little correlation between sample rate and audio quality through the USB cables? Surely, a poor USB cable with "slow" risetime should sound worse with 24/96 or higher bitrate music, right? Yet the majority of the subjective complaints focused on 'Hotel California' off Hell Freezes Over CD or Oscar Peterson's We Get Requests FIM K2 CD. Heck, even The Beatles' Abbey Road 24/44 barely taxes the USB interface. The only really "hi-res" track was the 24/192 Helge Lien Trio Natsukashi which was mentioned 3 times total in the whole subjective write-up. Surely, to get a sense of how well USB cables work, we need to grab some DXD and DSD128 material, right? If timing/jitter were that important, there could be big issues with DSD sampled in the MHz range, don't you think?*

6. In the subjective analysis conclusion, you see the following comments: the QED is described as "almost sounds louder", Transparent Performance's cable had "rich and warm sound proving a little too luxuriant at times", Wireworld's Starlight 7 had the "woomph of air that would normally accompany the deepest bass was subjectively filtered away", or the Crystal Cable's "greater extension" - why not measure those things? Surely you can easily detect amplitude changes, differences to frequency extension, tonal changes to add "warmth" in the DAC output, right? Yet, for the objective test all we get is a risetime measurement of a piece of wire and the subjective testing done through a second digital interface of unclear quality and unclear blind testing protocol. Therefore neither the subjective or objective results appear all that convincing.

Assuming we do acknowledge this article as having validity and not just an example of a majorly flawed study (needless to say, the setup using a digital coaxial cable is a BIG problem IMO), there is one thing we learned in this piece for sure that could be of value and a good reminder. Don't buy the £70 Vertere Pulse D-Fi USB cable - apparently it's constructed out of "twin coaxial cables" (err, impedance matching anyone?), has bad rise time (27.6ns), and sounds questionable with "softening of its extreme treble and loss of atmosphere". I wonder if connecting this cable to a hard drive might show slower transfer speed due to high data error rate. Evidence that you can spend more and get significantly less quality than the 'freebie'?

Assuming we don't believe the listening tests are valid because of the methodological issues, then perhaps this article serves as an example of the unreliability of subjective listening. Isn't it possible they're all just listening to the same sound with the "jittery" coaxial digital cable and interface and coming up with different impressions? Since we have no concrete idea of how the "blind test" was conducted (especially what level of statistical confidence we can expect), it's quite possible the term 'fantasy' would be appropriate to describe the results (but verboten).

Enjoy the music folks... IMO, there is still no good evidence that USB cables make a difference to sound (well, at least with decent modern gear!).

My musical selection tonight: C.C. Colletti Bring It On Home (HDTracks 24/96 Binaural recording). Wonderfully spatial sonics and details from the ASUS Essence One + Sennheiser HD800! Have a listen to that title track.

* For the record, I do not believe there is any reason to think DSD would be affected by USB cable jitter to an audible degree with good modern DACs. Just wanted to throw out a thought which the jitter-fearing-audiophile might bring up one day.

A number of moons ago (December 2012), this blog was started to obtain data on ~320kbps MP3 vs. lossless audio. Near the end of data collection for that study, I started performing some tests and began posting them on the Squeezebox forum for discussion.

Over the ensuing months, I started posting data on the family of Squeezebox devices which I collected over the years as the foundation of my sound system at home. Up to this point, I have had the opportunity to measure ALMOST the whole family of products from the SB3 onwards...

Squeezebox 3 / Classic

Obviously missing from this lineup is the Duet - a combination of the Receiver and the LCD-endowed Controller which I have always wanted to try but never saw a unit locally to buy back in the day. I want to send a big thank-you to 'fordgtlover' from the Squeezebox forum for taking the time and expense to send me the Squeezebox Duet for testing - all the way from Australia no less!

Here she is:

The Controller comes with a nice weighted metal stand which also plugs in as a recharger. The LCD screen is reasonably bright and easy to read. Buttons have a nice click and selection is made with the turn wheel. One could ask for a smoother feel to the wheel, but as is, it's reasonably functional. Even though you can connect the Receiver via ethernet, the Controller obviously needs a functioning WiFi network.

The Receiver is a relatively non-descript black box with just a lit wireless icon on the front to indicate power & device status. Getting it connected up was quite simple and quick using the Controller - see here for the details.

A look at the back ports on these devices. From left to right of the Receiver: analogue stereo, TosLink digital, coaxial digital, Ethernet (10/100), and wallwart power (9V, 0.56A). You can make out on the top of the Controller a phono plug as well for audio output to headphones on the go.

I: Receiver Measurements

Internally, the Receiver uses the Wolfson WM8501 which is spec'ed at 100dB SNR and although it can go to 24/192, the Receiver is limited at 24/48 (as are all the other Squeezeboxes except Transporter and Touch).

Okay, lets get into it. First I wanted to have a look at the analogue outputs of the Receiver. Since this is a loaner unit, I'll try to extract as much as I can before sending it on the long journey home to Australia...

Lets pull up an oscilloscope reading* - 1kHz 0dBFS square wave (24/44):

The yellow channel is the right, and blue left. Notice there is some channel imbalance on this unit with the left channel a bit louder. Left channel peak voltage measures at 2.47V, right channel 2.40V.

The square wave itself looks nice. Minimum noise, good contours with minimal ringing (compare with the Controller below). From this, one would suspect that the usual RightMark analysis would show good results.

Here's the impulse response (as usual, 16/44 impulse recorded with the E-MU 0404USB at 24/192):

A standard linear phase filter response. Absolute polarity is maintained. We can therefore expect a sharp roll-off at Nyquist.

A. RightMark results:

The usual setup:

SB Receiver --> Shielded RCA --> E-MU 0404USB --> Shielded USB --> Win8 laptop

Firmware: 77

Receiver volume 100%.

Cables were the same for all these measurements (a 3' shielded RCA cable from Radio Shack, shielded 6' USB cable).

I tried both ethernet and WiFi. Since I could find no difference, I'll just present the WiFi data here. I noticed that the WiFi in the Receiver was able to achieve better signal strength than the Touch at the same location. 2 floors up from the ASUS RT-N56U router, my Touch got ~65% strength, while the Receiver was up at ~75% (Controller also about 75%).

Where I did the measurements, the Receiver was up around 85% WiFi signal strength.

As you can see, I measured the Receiver against the Touch all the way up to the max sampling rate of 24/48. Also, I included the 24/44 data from the Squeezebox 3/Classic as comparison. The graphs below are derived from the 24/44 data. Notice that we gained 4dB from 16 to 24-bits using the Receiver... Not much.

Frequency Response:

Noise:

THD:

Relatively stronger odd-order harmonics up at 3, 5, and 7kHz compared to the Touch or SB3.

Stereo Crosstalk:

As you can see, a mixed bag. The Receiver's frequency response is similar to the Touch; both are flatter than the SB3 which tends to roll off the deep bass a bit more. The Touch clearly beats the other two in terms of noise performance (SB3 and Receiver about equivalent here), but scores lower with stereo crosstalk (not really an issue IMO). Total harmonic distortion was higher with the Receiver.

B. Dunn Jitter Test (16 & 24-bits):

16-bits:

24-bits:

Low jitter using this test in the analogue output.

II: Controller Measurements

The controller is a neat device. As shown above, there is a metal charging base and it's motion sensitive so will automatically wake when picked up (a few seconds wait time to wake up if not in the charger). It's pretty good at doing what it's meant to - as controller for the various Squeezebox units (not just the Receiver). The scroll wheel allows accelerated item selection, but these days, I mainly use a tablet to control the Squeezebox devices and it's of course quicker to type in search items than scrolling through letters.

The Controller has a standard 3.5mm phono plug up top for headphones. Although it can also play back audio, as far as I know, this feature was implemented at a "beta" level only. You have to go into Settings -> Advanced -> Beta Features -> Audio Playback to turn this on. Because the unit has a little speaker inside, I also needed to download the "Headphone Switcher" app from Applet Installer which adds an item in the "Extras" menu to turn the headphone on. Eventually I was able to get the unit to automatically switch between the speaker and headphone after some fiddling and rebooting the device (honestly I don't know which step was the key - the nature of 'beta' I guess). I did notice some slowdown to the UI responsiveness when streaming audio to the Controller.

According to the SB Hardware Wiki, the Controller has a Wolfson WM8750 inside. Understandably, this is a low power chip meant for portable devices. Data sheet lists DAC SNR up to 98dB. Maximum sample rate goes up to 24/48.

Setup:

Controller --> phono-to-RCA shielded Radio Shack cable (3') --> E-MU 0404USB --> shielded USB --> Win8 laptop

Firmware: 7.8.0-r16739

Controller volume 100%.

Here's what a 1kHz square wave, 0dBFS, 24/44 looks like under the oscilloscope playing off the phono plug:

In comparison to the Receiver above, it's clear that this isn't going to be "hi-fidelity". There's a bit of ringing and noise evident. Peak voltage is around 800mV and the two channels are reasonably balanced.

Here's the impulse response:

Upside down - the Controller inverts polarity. Standard linear phase digital filter.

A. RightMark results:

Comparison is made with the Receiver and Touch. Numerically, the thing that really sticks out is that massive intermodulation distortion up at 4%! I'm a bit surprised how the program did not calculate a higher THD as well given how nasty the graph looks - this is why I post up the graphs as well... I checked and double checked the testing to make sure settings were not clipping the signal. Made no difference - it is what it is!

No meaningful improvement with going to 24-bits.

Frequency Response:

Odd noise spikes from 5kHz up. Have not seen this before in my other tests. I did the test 3 times and this appears to be a real finding. Also, earlier roll-off down to -3dB by 20kHz.

Noise:

THD:

This looks bad but the score was about the same as the Receiver. RightMark is probably just looking at the odd and even harmonics for the calculations.

Intermodulation Distortion + Noise:

Ouch. Non-linearity is an issue.

Stereo Crosstalk:

As I said earlier - the Controller isn't high fidelity :-).

B. Dunn J-Test:

16-bit

24-bit

Very strong jitter! Many many nanoseconds of jitter :-). Again, this is not distortion from clipping.

III: Summary (Analogue Output)

Receiver: Reasonable analogue output from the device. It performs similar to the old SB3 in terms of dynamic range and noise floor. I hooked up the Receiver to my bookshelf system upstairs ('vintage' Sony MHC-1600 from university days powering some Tannoy mX2's) for a few days to listen. Very enjoyable - got a chance to listen to the Into Darkness soundtrack, the recent HDTracks release of Cole Español, and stroll down pop memory lane with Samantha Fox's Greatest Hits :-). I pulled out the Squeezebox 3 to compare and subjectively, I agree that there's a bit more bass with the Receiver and Touch than the SB3. While that channel imbalance was measurable, I didn't find any gross anomaly with the speaker system but I could hear the difference with the AKG Q701 headphones for example listening to where Ella Fitzgerald's voice was centered on Ella Sings Gershwin, I'd say it's subtle so would not affect my listening pleasure (could also be placebo since I was specifically listening for this!).

Controller: Firstly, how does it sound? Well, perhaps surprisingly OK :-). It's low powered so I listened with a pair of JVC HAFXC80 IEM headphones. I would compare the sound output to what I hear off my Samsung Galaxy S2. If you have a Controller, have a good listen - that's what a high jitter device sounds like (I have not ever seen this many side bands in all the testing so far!). For me, the best I can describe is that the Controller sounds "distant" even with volume pushed up and slightly "hollow" compared to listening through the headphone out from the Touch. The percussion at the start of Star Trek Main Theme from the "Star Trek: Into Darkness" soundtrack for example sounded less defined and spatially smeared, getting worse as the orchestral dynamics build up as if there's some low-level static in the background. How much of this can be definitively attributed to the high jitter is hard to say since the unusual frequency response, noise levels, intermodulation distortion and headphone output limitations are all significant factors in the sound. I highly recommend just streaming 192kbps MP3 as this will improve the responsiveness of the Controller and you're not going to hear a difference.

"fordgtlover" wanted to know how well the Receiver serves as a digital transport - good question! This then will be the topic for the next installment along with comparisons with the Touch... Stay tuned... Something tells me this is going to be complicated and hopefully quite interesting!

Using suggestions from this page, the Touch can be used to transport DSD to the TEAC as DoP wrapped around a 24/176 FLAC file through Triode's USB kernel. Neat hack - proof of bit perfect transmission. Unfortunately the files are huge, so I likely will await an efficient solution. Note that this is NOT the test setup described below, just something cool. :-)

Let's talk about digital transports for a bit.

Most of us I'm sure remember those days when the only way to get digital data to an outboard DAC was through a CD transport. Although we can still resort to a CD reader, I suspect that many of us here have gone mostly into the computer audio realm with data on hard drives or flash/SSD devices. Thankfully gone are the days when the data being read off the CD could be inaccurate and interpolation may be needed in realtime, or be susceptible to mechanical failures of CD drive mechanisms (though hard drive failures and need for backups present another challenge).

I have shown that bit-perfect data can be transferred and played back without any concern off a USB asynchronous interface (eg. the various Mac and Windows software players and laptops). While I have the Squeezebox Receiver still on hand, let us have a look at the effect of using different transport devices with S/PDIF interfaces objectively.

Remember to keep in mind that the S/PDIF interface, unlike packetized asynchronous USB (or ethernet) conducts its data transfer in a unidirectional serial fashion formalized around 1985 when the AES3 (AES/EBU) standard was also laid down. As I think many of us have read, S/PDIF combines the data and clock signals using "biphasic mark code" and it is this "feature" of the interface that has resulted in many an audiophile nightmare regarding timing issues - jitter concerns especially well publicized. This is to a large part the basis for the well known paper by Dunn and Hawksford in 1992 asking "Is The AESEBU/SPDIF Digital Audio Interface Flawed?". (Of course the topic of jitter can be very complex as described in the paper going far beyond what we need to concern ourselves with here.)

With that said, let's be practical and see what the results looks like with the different devices using TosLink and coaxial interfaces (with comparison to asynchronous USB)...

Setup:

For this round of testing, I decided to take a break from the TEAC UD-501 DAC and go back to the ASUS XONAR Essence One for a bit. Although I have been listening to the TEAC a lot in the last couple months, on a daily basis, I still use the ASUS Essence One at my computer workstation and it was just more practical to run these tests there. Despite my concerns around the upsampling feature of the ASUS, it measures well and sounds excellent.

Here's the hook-up:
* Transport device * -> Coaxial/Toslink/USB cable -> ASUS Essence One -> Shielded 3' RCA -> E-MU 0404USB -> 6' Belkin Gold USB -> Win8 laptop

Coaxial cable = 6' Acoustic Research
TosLink cable = 6' Acoustic Research
USB cable = 6' Belkin Gold

Transport devices tested:
1. Squeezebox 3 -> Coax / TosLink
2. Transporter -> Coax / TosLink
3. Receiver -> Coax / TosLink
4. Touch -> Coax / TosLink
5. Laptop -> CM6631A Async USB to S/PDIF -> Coax / TosLink
6. Laptop -> Async USB direct to ASUS Essence One

As you can see I've got the host of Squeezebox devices on the test bench along with the usual two ways to connect the computer's USB port to DACs (direct or through USB-S/PDIF converter).

I. RightMark Analysis:

Since there are so many devices/combinations, I'll show the results a few at a time to demonstrate what was found. Let us start with the results of the four Squeezebox devices. I decided to "max out" the capabilities of the SB3 and Receiver by using 24/48 sampling rate:

Numerically, not much difference... From my subjective listening during the tests, I would agree with these numbers in saying that "it sounds like the Essence One DAC"; there are more similarities in the sound than subjective differences.

Let's have a look at the frequency response graph because there does appear to be some difference - here's coaxial interface only:

Hmmm, interesting! Small differences at the top end. Let's zoom into that top end and have a good look:

Notice the shape of the curves suggest slightly earlier roll-off with some devices. The flattest, most extended frequency response (and possibly most "accurate") is the Transporter, followed by SB3, then Touch, and Receiver. Remember, we are talking only about 0.15dB difference between the Transporter and Receiver at 20kHz; not perceptible IMO but since we're looking for evidence of a difference, useful to note.

Let's now include the TosLink measurements:

Even though we ran out of colors, it doesn't matter because there are still only 4 curves. There is no difference between coaxial and TosLink; they overlay on top of each other essentially perfectly.

Let us now add the computer-USB interfaces (take away the TosLink since no difference):

As you can see, the flattest response curves come from USB direct and Transporter. Here's the ranking: Transporter, ASUS USB direct, SB3, Touch, CM6631A, and Receiver. We'll talk more about this later, just keep in mind then that frequency responses are *slightly* different between transports despite bit-perfect settings...

The rest of the RightMark graphs - no significant difference:

II. Jitter Analysis:

Dunn J-Test stimulation of jitter. To keep it more manageable, I'll group them into 16-bit and 24-bit side-by-side first, let's just look at the coaxial interface here:

A. Squeezebox 3 (16-bit / 24-bit):

B. Transporter (16-bit / 24-bit):

C. Receiver (16-bit / 24-bit):

D. Touch (16-bit / 24-bit):

E. Laptop -> CM6631A USB to coaxial (16-bit / 24-bit):

F. Laptop -> USB direct (16-bit / 24-bit):

Objectively, it looks like the CM6631A USB-to-S/PDIF and USB direct 24-bit graphs are cleaner, and of the Squeezebox devices, the Transporter on the whole seems to have the least data-correlated jitter. Even at its worst, the sidebands for the SB3 around the primary signal is down around -120dB. Is this a problem? I doubt it since auditory masking will easily make this inaudible (assuming one could even hear down that low around 12kHz pitch). Furthermore, in theory, the J-Test should create a "worst case scenario" for jitter which is unrealistic in real music.

Here's the difference between coaxial vs. TosLink:

A. Squeezebox 3 24-bit, Coaxial vs. TosLink:

B. Transporter 24-bit, Coaxial vs. TosLink:

C. Receiver 24-bit, Coaxial vs. TosLink:

D. Touch 24-bit, Coaxial vs. TosLink:

E. CM6631A USB to S/PDIF 24-bit, Coaxial vs. TosLink:

In general, we can say that indeed TosLink is worse (remember however TosLink is immune to electrical noise with galvanic isolation so there are some positives in this regard). Interestingly this is very clear with the Transporter! However, increased jitter with TosLink is not a given because the SB3 and Receiver seem to behave in the opposite fashion and show less jitter artifacts with the TosLink interface.

Remember that all of these jitter graphs are indicative of the interface between the transport device connected to the Essence One DAC. The graphs could be different with another DAC since much of the result will depend on the accuracy of the DAC in extracting the clock information and what other steps it might take (eg. data buffering, reclocking) to further stabilize the timing - it's not just about the transport device.

III. DMAC Protocol

So, up to now we can see differences between bit-perfect devices with RightMark, and obvious differences with the J-Test. How do they sound? Let's see what the computer "hears". For this test, I am using the Transporter playback as reference against which all the others are being compared.

First, I must admit that I'm not as confident about these numbers as I am of the graphs and plots above simply because it was really tough getting this done properly! From previous experience with the Audio DiffMaker program, results can vary depending on environmental factors like temperature of the equipment and subtle "sample rate drift" over time. With each transport measured, cables needed to be reconnected, settings needed to be changed, and for each condition, I ran the test 3 times to get a sense of the "range" of results. Admittedly, I made an error with the 'USB direct' measurements and did not realize this until after the fact so did not include the results here (foobar was accidentally set to output 16-bit instead of 24-bit).

The bottom line is that the results suggest that each device "sounded" different according to the computer. Instead of the usual high "correlated null depth" like in my previous tests with player software around 80-90dB (similar to the Transporter tested against itself above), we're seeing numbers in the 60-80dB range between transports. The computer thought the Squeezebox 3 sounded the most different from the Transporter. Good to see that it was able to detect the Receiver playing 320kbps MP3 as "most different" (ie. lowest correlation) to provide a point of reference. A reminder, this measurement is logarithmic so the actual mathematical difference between the MP3 sample compared to the others is larger than what it might look like on the graph.

Remember that this is a measurement of the difference between each device and the Transporter connected to the Essence One. There is no implication here of whether one sounds "better" than another since that would of course be the listener's subjective judgment call.

IV. Summary

Let me see if I can summarize this based on the results here along with what I know/believe over the months of testing as applicable... Q&A format:

Q: Do all bit-perfect transports sound the same?

A: Based on the results, not exactly. Even though bit-perfect (I have verified this with the Touch, Transporter, SB3, CM6631A, ASUS USB direct with ASIO), small differences in frequency response can be measured. Furthermore, jitter analysis clearly looks different between devices and this also varies between coaxial and TosLink interfaces (with TosLink generally worse than coaxial for jitter). Likewise, the DMAC test also suggests the level of audio correlation when playing musical passages is not as high as previous tests with bit-perfect software or decoding lossless compression. Within the Squeezebox family, not surprisingly the Transporter performed the most accurately with flattest frequency response and lowest coaxial S/PDIF jitter, although I was quite surprised by the stronger TosLink jitter.

Q: Why do you think the frequency response varies?

A: My belief is that this is not a jitter issue. The reason I say this is that there appears to be no difference between coaxial and TosLink even though jitter varies between the two interfaces as demonstrated by the Dunn J-Test. I believe that this is the result of mild clock speed / data rate differences of the transport devices. Since the word clock has to be recovered from the S/PDIF signal, clock accuracy is dependent on the transport's internal clock - some transports may be timed a little quicker, some a littler slower and the DAC has to adjust to this (of course the E-MU 0404USB ADC measuring the audio has a part to play in setting where it believes the roll-off should be). This frequency roll-off variability is not seen with laptops connected to an asynchronous USB device for example (that's of course the point of being asynchronous; not time-coupled to the data sender by having the recipient working off its own clock and telling the sender to speed up or slow down if necessary).

Q: But surely different/better/more expensive digital S/PDIF cables can help?

A: No. I don't think so. As I have measured and discussed before, digital cables make no substantial difference to timing/jitter as far as I can tell. Even though very long or poorly constructed cables may add to the jitter, the difference IMO is much less than what I'm showing here and as far as I can tell is irrelevant for a reasonable length of decently constructed coaxial/TosLink.

Q: Those jitter plots look nasty... I bet I can easily tell the transports apart!

A: Of course, anyone can claim anything over the Internet or in print since there are rarely if ever any actual "double checking" with sound methodology or formal peer review in the case of print magazines (obviously these are not scientific journals). Although I have shown these measurable differences, as a (currently) 41 year old male who works in an office environment, have generally avoided very loud concerts, and have a hearing frequency threshold around 16kHz, I do not believe I would be able to differentiate any of these bit-perfect transports in controlled testing with the same ASUS Essence One DAC.

Q: Surely you just need better gear to hear it!

A: The data correlated jitter with any of these devices would be >100dB below the primary signal. The frequency response difference is less than 0.15dB at 16kHz (my frequency threshold). Unless there's some significant interaction that causes anomalies in the output significantly beyond what I measure here, these difference would be inaudible to me irrespective of the quality of the sound system. Of course if you have younger ears and better hearing, this could be different. I believe speakers and headphones would introduce much more distortion and change to the frequency response than what I'm measuring here with a good modern DAC.

Q: Well, if that's the case, then I might as well go for the cheapest digital transport/streamer I can find, right?

A: Well, maybe, maybe not. When it comes to sound quality, I think a digital transport would have to be quite incompetent to sound poor (eg. non-bit-perfect, horrific jitter or imagine if the frequency response rolled off way too early because of severe S/PDIF timing inaccuracies). Therefore, spending more on a digital transport is IMO not primarily about sound quality but rather features and the aesthetic "look and feel" you're after (eg. better remote, can handle higher sampling rates, more reliable, fits into the decor...). Sound quality IMO is better served by putting the money into good speakers/room treatments/amp/DAC. Back in the "old days" of CD spinners, better mechanics with higher reliability and accuracy just cost more money. Even then it's not a given; I remember spending five times more on a higher model Harmon/Kardon CD player 20 years ago and that failed within three years whereas an inexpensive JVC from Costco with digital out still runs fine today. I have not had occasion to try the "low end" devices like the <$100 media streamers (eg. WD TV) to see how those compare to the Squeezebox dedicated audio units.

BTW: If you're not aware, the Squeezebox devices by nature are asynchronous since they receive the data through WiFi or ethernet from Logitech Media Server and buffered with a decent amount of internal memory. You can see that the TosLink and coaxial connections have worse jitter than what's measured directly off the analogue outputs (eg. look at SB3, Touch, Transporter, Receiver jitter measurements).

As usual, feel free to comment or link to any good data you may have come across regarding this topic especially if conclusions are different from what I've presented.

Musical selection this evening:
Philippe Jaroussky - Carestini (The Story of a Castrato) (Virgin Classics, 2007) - amazing vocals and fascinating musical history. ("It's a man, baby!" -- Austin Powers)

Happy listening! ;-)

Well, after finishing the last couple of posts, I sent the Squeezebox Receiver back to fordgtlover. Thanks for the opportunity to test out another member of the SB family! I very much appreciate the generosity!

Before the Receiver left on the long trip back to Australia; I decided to take a family portrait:

I remember patiently awaiting the arrival of the Slim Devices Squeezebox 3 back in late 2005 after putting in the pre-order. It's thanks to this little device that my audio listening habits changed forever away from physical media. Despite the ups and downs with various versions of the server software, these little boxes have certainly made access to my own music remarkably easy in a way almost unimaginable previously. My father has access to my music library across the city through his Touch and wherever I go in my travels, music streams to the PC/Mac as well. The stability of my server these days with >5000 albums is fantastic with uptimes of months (this is with Windows, and the only reason it goes down these days is because I install Windows updates every few months).

Sadly, the Squeezebox line was discontinued in August 2012 by Logitech.

Looking back, I'm still impressed by the technology put together by the Slim Devices team to get the infrastructure in place, and later with the Logitech team and especially the Touch. Even today, the objective analogue audio output performance of the Transporter remains superb (it came out in September 2006) and as I look at all that can be done with the Touch (came out April 2010) including the ability to handle 24/192 (EDO kernel), communicate with a number of USB DACs, transport DSD64 via DoP, it's amazing the power of what can be done with an Open Source architecture where the "community" is empowered to maximize the machine's capabilities. Have a look at the used prices of the Touch these days, it's a reminder of just how much value it offered audiophiles. The computer audiophile community needs more devices of this caliber; devices that can "raise the bar" and do it without need for valuation as a luxury or "artisan" item when it's just "good sound" and "good functionality" that I believe many of us are after.

Looking ahead, it's great to see ongoing development in the Squeezebox community towards replacement hardware and software - check out the communitysqueeze.org FAQ. Wonderful to see the DIY machines being put together out there (check out the picture gallery!). With the availability of small, low powered, but reasonably fast computers like the Wandboard and the already ubiquitous ports of client software like Softsqueeze/Squeezeplay/Squeezeslave, Squeeze Player (Android), Squeezecast (iPod/Phone), I have a feeling that I'll be running my Logitech Media Server (or some equivalent) for the forseeable future...

... Now, if they could support multichannel FLAC/WAV and native stereo/multichannel DSD (I still think there's a need for better file formats than .dsf and .dff), I think we'd have all of audio covered. :-)

Vive le Squeezebox!

----

Well, it's summer time in the northern hemisphere and that means the kids are out of school. The heat wave finally arriving here in Vancouver. Time for some vacation, BBQ's, camping, lazy afternoons, general R&R, and of course good summer tunes (Katrina & The Waves I'm Walking On Sunshine is playing in the back currently). :-)

I'm also starting to go through Ethan Winer's recent 2012 book "The Audio Expert: Everything You Need To Know About Audio". Only started and already it's a good read written in an accessible manner for those interested in the science and technical aspects of audio without going deeply into the math. The Kindle edition (IMO a bit pricey) along with some mindless fiction like Dan Brown's Inferno will likely accompany me on the summer trips coming up.

As always, enjoy the music everyone!

According to some... CD sound quality is inadequate.

In the comments to a previous post, Fabio Zolli asked:

I would like to ask your opinion about high resolution audio files like 96/24 or 192/24 (I know it's off topic but I didn't know where to post). I think there isn't a really audible improvement among standard 44/16, taking into account also your mp3 test. Thanks for your attention.

Thanks for the note Fabio.

The MP3 test shows that even if we do hear a difference, at the level of quality tested (~320kbps MP3), without comparing the "studio master" sound, it's virtually impossible to know which is better/more accurate. Amazing given just how much data is being thrown out by MP3 encoding!

Personally, up to now I have not been able to ABX a difference between properly dithered 16/44 and the 24/96 source. Others have also done some fantastic testing around this like Mitchco at Computer Audiophile with different test methodologies.

It has been 30 years since the introduction of the CD (PCM 16/44) standard for mass consumption. The fact that we have not seen good controlled tests to show that 16/44 is somehow "lacking" (when listening to music at normal volumes) compared to 24/96+ or SACD tells me that 16/44 is most likely good for any circumstance. Much debate for example was sparked in 2007 by Meyer & Moran when they published their report based on work with the Boston Audio Society when listeners could not detect a significant difference between hi-res (DVD-A and SACD) analogue output versus the output from a 16/44 A/D/A loop.

Knowing the above, reading stuff like this should make an unsuspecting audiophile wonder why the inferiority of 16/44 wasn't proven ages ago! The author writes: "For me its clear that 24/96, 24/192 and DSD are superior to 16/44.1 in many meaningful ways..." Really?

The only rational issue I have come across "against" the 44kHz sampling rate is that with older DACs, the Nyquist frequency at 22kHz is too close to the potentially audible 20kHz upper range and the steep "brickwall" filter may cause audible effects. But this issue was addressed with upsampling and slower roll-off filters long ago (I know... yet another contentious issue for some).

Admittedly, I will seek out 24/96 audio for favored albums and rip my SACDs as 24/88. My rationale's simple... Since storage is cheap these days, I consider doing this as reasonable "insurance" to guarantee that everything that science tells us is humanly audible is captured in the digital sampling with plenty of headroom above 20kHz. I believe this is a reasonable price to pay in terms of storage even if I do not ultimately get an audible benefit because my ears aren't good enough for the task (like one pays for insurance never knowing if it's actually needed). Furthermore, objectively, all my DACs seem to be optimal at these 2x sampling rates as well showing nice frequency extension and excellent dynamic range measurements.

As for 24/192... Christopher "Monty" Montgomery at xiph.org has written an excellent review of this topic including examples of how this could be detrimental to sound quality (like intermodulation distortion which also can be shown with 24/96). Likewise, Dan Lavry has written about the technical issues of 24/192 in his whitepapers like this one.

Personally, I have yet to find music I thought could benefit from this high sampling rate (how many microphones can even accurately record >40kHz?). Even though storage is cheap, I'm not convinced that there's anything to be gained going from 96kHz to 192kHz theoretically or otherwise. The cost-benefit ratio is hard to justify when benefit seems to be zero! (BTW I have come across some albums like Carmina Celtica from Canty off the Linn download site at 24/192 with unusually high noise level above 45kHz which I suspect could create some very nasty intermodulation distortion if not properly filtered.)

Bottom line:
I'm happy with 16/44 if the music is well recorded and mastered. For my favorite tunes, I'll go for 24/88 or 24/96 if available to "ensure" that I'm not missing something. To date, I have happily downsampled many 24/192 albums to 24/96 (or 24/176 to 24/88) without any reason to think that I'm somehow "missing out" (rather than gaining space for more music!). As for DSD, I only have a few albums verified to be sourced from an original DSD recording.

As many others have commented before, the main "enemy" to good sound these days is not about the audio format - yes, even the much-maligned-by-audiophiles MP3 can sound excellent if the underlying content is good. Rather, the way the album was recorded and mastered is more important. For example, the loudness war has caused more damage to sound quality than we can ever gain going from 16/44 to any high-resolution format. I believe sound quality "evangelists" like Neil Young would do well to "wage heavy peace" on that silly "war" and in the process stay relevant.

Feel free to leave me a note especially if there are good reasons to consider keeping 24/192 music on my server system! :-)

One more thing... There is one situation where you'd definitely want to either upsample or run a higher sampling rate - if you're using a NOS DAC. However, this is more to do with reducing aliasing distortion in the audible spectrum.

Musical selection tonight: Going to have a listen to k.d. lang's Ingénue (1992) again... It's been awhile!

The sentence says "supported by Japanese SACD manufacturer" (whatever that means!). An example of how the term SACD gets thrown around in cheap domestic and pirated Asian markets (this wasn't a true XRCD either)...

As I mentioned previously in my post on SACD (and DSD), there are a number of SACDs I have digitally ripped over the years that appear to be sourced from 44kHz PCM. This is of course the same sample rate as good ol' RedBook CD and therefore it's unlikely that these titles should sound any "better" than the CD release since the PCM-to-DSD conversion process will add some distortion to the original signal.

It's not difficult to detect these releases because of the "brickwall" loss of frequencies beyond 22kHz. Note that this list is of course unofficial and even though there's evidence that these come from 44 or 48kHz PCM, it's still possible they're from 24-bit data which could still be "hi-res".

An example of the 22kHz brickwall - Thelonious Monk's "Between The Devil And The Deep Blue Sea" off Straight, No Chaser SACD. Notice the typical ultrasonic 'noise shaped' SACD quantization noise from 22kHz up - filtered off in this case before 40kHz.

The reason I decided to post this list up was because even to this day, some folks will use questionable albums such as these to prop up the putative superiority of the SACD format or compare the DSD rip with the CD layer... Even as recently as the May 2013 edition of Home Theater magazine - David Vaughn reviewing the Marantz AV8801 & MM8077 spoke of how he preferred the two-channel SACD layer of Norah Jones' Come Away With Me (p.41). There is of course one "inconvenient" problem; we've known since 2004 that this SACD was in fact a RedBook CD upsample! Therefore, the preference must be the result of perceptual bias, euphonic distortion by the DSD conversion, or maybe his SACD player is poor at playing CD. I have seen these kinds of comparisons and biases based on resampled SACDs made over the years both in the print magazines as well as on-line (eg. see this review) and in forums.

By no means do I believe this is a complete list - just ones I've run into over the years or confirmed by friends. I have tried to comment on these SACDs as part of my reviews on places like SA-CD.net but notice that these comments tend to get censored over time - I guess I can't blame them since they have something to sell :-).

Similar situations exist in the DVD-A world especially with upsampled 24/192 material (alas I never kept track of these but remember Frank Sinatra At The Sands was likely upsampled).

In alphabetical order of artists (threw in a few spectral screenshots):

Albert King - I'll Play The Blues For You (2004 release)

Albert King & Stevie Ray Vaughan - In Session (Probably 48kHz upsample, 2003)

Andrew Lloyd Webber - The Phantom Of The Opera OST (2004 Joel Schumacher movie)

Babyface - The Day (2001 SACD)

Beoga - Live At Stockfisch Studio (2010)

From "Factory Girl"

Blue Öyster Cult - Agent Of Fortune (2001 release, both stereo & multichannel layers likely from 44kHz)

Bryan Ferry - Frantic (both stereo & multichannel layers likely from 44kHz)

Cat Stevens - Tea For The Tillerman (2011 Analogue Productions - surprising! Maybe 48kHz source)

From "Wild World"

Celine Dion - A New Day Has Come (2002 - no surprise with pop albums, maybe 48kHz source)

Cowboy Junkies - Whites Off Earth Now!! (MFSL 2006 release - originally PCM recording)

Dead Can Dance - MFSL SACD Box Set (2008)

From "The Carnival Is Over" on Into The Labyrinth

Dire Straits - Brothers In Arms (2005 release - well known PCM recording)

From "Money For Nothing"

Donald Fagen - The Nightfly (2011 release - well known PCM recording, DVD-V version better IMO)

Elvis Presley - Elvis Is Back (2012 Analogue Productions - maybe so old there's no high frequencies!)

Eugene Ruffolo - Santa Sings The Blues (2009 release, Stockfisch)

Joe Satriani - Engines Of Creation (2000)
- Also, the CD layer has DR7 vs. SACD stereo layer with DR11. Different mixes with very dynamically compressed CD layer!

John Eliot Gardiner & Philharmonia Orchestra - Grainger: The Warriors & Holst: The Planets (2003)

Keb' Mo' - The Door (2000 release)

Ken Ishiwata's Band - Marantz: Ken Ishiwata's 30th Anniversary (2009)

Nick Drake - A Treasury (2004)

Norah Jones - Come Away With Me (2003)

Peter Gabriel - So (2003 release - I suspect 44kHz recording but transferred from magnetic tape to DSD, thus the tape bias in the high frequencies and higher noise)

"Sledgehammer"

Peter Gabriel - Up (2003 release)

Pixies - Bassanova (2008 MFSL)

Ryan Adams - Rock N Roll (stereo layer looks sourced from 44kHz, multichannel mix seems OK)

Sarah Brightman - Eden (both stereo & multichannel, stereo layer DR7 seems even more dynamically compressed than CD DR8)

Sarah Brightman - La Luna (like above, stereo SACD layer DR8 vs. CD layer DR9)

Simon Rattle, Libera & Berliner Philharmoniker - Tchaikovsky: The Nutcracker (2010)

Suitcase Pimps - Love Is Grand (2003)

The Dave Brubeck Quartet - Time Out (Columbia/Legacy & 2000 SME Japanese stereo layers)
- Note the multichannel Columbia/Legacy layer seems to be OK.

From "Take Five" original Columbia/Legacy SACD.

Thelonious Monk - Straight, No Chaser (1999 release, see image above)

Tony Bennett - Playin' With My Friends (2001 release)

Uriah Heep - Magic Night (2004)

Vince Guaraldi Trio - A Boy Named Charlie Brown (2004 release, 48kHz source?)

Vince Guaraldi Trio - A Charlie Brown Christmas (2003 release)

Yo-Yo Ma, Edgar Meyer & Mark O'Connor - Appalachian Journey (2000)

Yo-Yo Ma - Soul Of The Tango (2003 release)

Interesting how the list includes "audiophile" labels and even demo/promo material like the Ken Ishiwata Marantz SACD. I think it also doesn't help the SACD cause that early Sony SACDs like the Brubeck and Monk are among these. I estimate about a quarter of the SACDs I've had a chance to evaluate consisting of eclectic rock, classical, and jazz disks are of the suspected-44/48kHz-upsampled variety.

Remember that I'm not saying these SACDs are bad sounding, just that in principle, the CD layer could be a more accurate representation of the music since there's no PCM-DSD conversion going on. In fact, many of these sound great and suggests that CD quality may be good enough. Unless there's a multichannel mix that you want on these disks or the mastering is somehow different (eg. less dynamic compression), IMO, there's no point paying extra for these SACDs.

Feel free to drop me a note if you know more about the releases (eg. can confirm if 24-bit source) or if you know of other upsampled disks.

Musical selection tonight: On an 80's kick right now. Going to have a re-listen to Midnight Oil's Diesel And Dust to get into the weekend mood... First pressing from 1987 of course. :-)

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

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

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

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

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

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

RightMark results:

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

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

Frequency Response:

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

Noise:

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

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

THD:

Another view of the ultrasonic noise.

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

If we compare it to the PCM:

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

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

PCM 24/192 FLAC played back:

DSD64 KORG transcoded DFF file played back:

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

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

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

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

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

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

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

When it comes to technological "toys", I've vacillated over the years between the accumulation of digital photography gear and audio stuff... As I'm sure many of you know, "pixel peeping" is the act of "using 100% crops and similar techniques to identify flaws that have no effect on the photograph under real-world conditions" (Google web definition). Back in the "old" days (like a decade ago), the act of pixel peeping wasn't all that unreasonable since the differences visible could be demonstrated on photo-enlargements. When I was using my old Nikon D70 with 6 megapixels, sharpness at the pixel level was a significant consideration with moderate enlargements like 13"x19"; imperfections like moiré could be seen in the final product as well. Monitor resolution wasn't that high back then either so fine details were easily obscured.

Fast forward these days and I'm now using the Nikon D800. At 36 megapixels viewed on a >2MP monitor; unless I'm printing huge enlargements, there really is little need to "zoom" down into the 1:1 pixel level to appreciate a high quality image... Sure, sometimes it's just fun to see how much detail has been captured especially when evaluating different lenses or to show off each hair follicle, but for the most part, "pixel peeping" has become quite unnecessary.

Although in daily usage, one might not need to "peep" anymore, if one were to publish camera body or lens reviews, any reviewer these days "worth their salt" would run the images through objective tests; including highly detailed "pixel level" tests or compare 1:1 images between cameras or lenses. Dynamic range, ISO-noise interaction, color accuracy tests, distortion characteristics (for lenses), effect of file formats (JPEG vs. RAW) of course all serve to complete the evaluation. The quality is so high these days among high-end cameras (SLR's, medium format digital backs...), it is with these detailed tests that we can fully appreciate the qualitative differences between top contenders. Subjective opinions in terms of the camera's touch-and-feel and user interface are important of course, but if you care about the potential image quality that can be captured, then objective tests are really really important. If you haven't already done so, just have a look at the camera reviews on www.dpreview.com and see how much work actually goes into what I respect as proper reviews of well engineered equipment! Also of interest, Hasselblad is trying to market "exotic" cameras at high prices by appealing to aesthetics (just look at the responses to see how people feel about that!). [Here's another one.] Is this what happens when technology matures and companies have difficulty competing on primarily technological merits?

I've often wondered why in the audio world, objective measures have so often been left out as part of the review process - especially as it comes to line-level devices like DACs. Maybe it's because digital audio matured earlier and we're going to see the same outcome with cameras one day. Around some forums, the mere mention of objective measures seems to be scoffed at - as if objectivism with audio gear is either "obsolete" or the sole domain of "high end" manufacturers with arcane tests out of reach of mere mortals. I know I'm digressing into "MUSINGS" territory here, but IMO, a good review needs to dig into the gear's objective properties so the reader can truly appreciate how it compares with other similar gear in order to have an informed opinion and gauge value as a (hopefully) well engineered piece of technology... Let's get back on track then with some "MEASUREMENTS".

For me, one of the most interesting "waveform peeping" tests consistently done by Stereophile over the decades on digital gear has been the undithered 1kHz sine wave test at -90.3dBFS. This is one of the most "microscopic" tests of DAC performance. It's simple and the result basically answers the question "can this DAC accurately reproduce the least significant bit (LSB) in a 16-bit audio signal?" At a glance one can tell at least 3 things:
1. Is the DAC "bit-perfect" down to that last 16th bit? (Assuming everything upstream is set up properly, you should see something resembling the 3 quantization "steps".)
2. Is the dynamic range at least 16 bits? If not, the waveform becomes obscured by excessive noise.
3. Are there anomalies to the waveform morphology to suggest "DC shifts" leading to "tilting" of the waveforms (power supply related issues). (For a good example of 60Hz low frequency noise effect, see the measurement of the Philips CDR880 Figure 7.)

The Stereophile website archive provides some lovely examples of this test dating back to the late 1980's such as the Philips LHH1000 from 1989 (check out Figure 5). How about the Naim NA CDS from 1992 (Figure 6) or the $8000 Mark Levinson No.35 DAC from 1993 (Figure 6, still not good). By 1995, we saw excellent performance like with the $9000 Krell KPS-20i (Figure 5). In a few years, by 1998, reasonable priced gear like the California Audio Labs CL-15 CD player was capable of similar accuracy at the $1500 price point. Since the millennium, this level of performance can easily be achieved within the $1000 price point and below (eg. the Rega Apollo from 2006). These days, the little Audioquest Dragonfly can do a reasonable job USB-powered at <$250 retail.

On the whole, this test has demonstrated the progression of improved accuracy over the years. State-of-the-art DACs like the MSB Diamond DAC IV (Stereophile October 2012, not on website) and Weiss DAC202 (Figure 6) are great examples of what this level of accuracy looks like (as opposed to expensive gear of questionable technical ability which I will not mention). IMO, well engineered CD/DVD/SACD/Blu-Ray/DACs these days claiming to be "high resolution" really should pass this test without issue. Nonetheless, there are recent devices apparently incapable of a low noise floor for whatever reason (eg. Abbingdon DP-777 Figure 15, surprisingly the recent Wadia 121 Decoding Computer Figure 6 didn't fare too well either).

I was curious whether I could run a similar test using my simple test gear... After all, so long as the DAC and measurement device can achieve >16-bits dynamic range reliably, one should be able to obtain a reasonably good set of measurements. So far, from what I've seen in the other tests, I should be able to reproduce this test with the E-MU 0404USB!

Here goes... Setup and procedure for the various DACs/streamers:
Test DAC --> shielded RCA --> E-MU 0404USB ADC --> shielded USB --> Win8 laptop

- I created an undithered 1.1025kHz sine wave at -90.31dBFS at 16/44. This is what an "ideal" waveform would look like with the usual Gibbs phenomenon (ringing) due to bandwidth restriction.

- Green is LEFT channel, Blue is RIGHT channel. Notice the phase inversion between the channels.

- For comparison, I also created the equivalent at 24-bit quantization:

- Capture the above at 24/88 with Audacity using the E-MU 0404USB. From previous tests, the E-MU functions very well at 2x sample rates (88 & 96kHz) with optimal dynamic range. Although not as good as a high precision oscilloscope used by Stereophile, this should be adequate to allow relative comparisons between different DACs. I used the analogue preamp on the E-MU to boost the signal by about 18dB to give me "more" amplitude to capture.
- As you can see above, I decided to plot the channels overlaid and inverted to compare precision of timing and amplitude.

Here are the results of this test on the various DACs I have around here:

TEAC UD-501 [2x BB PCM1795 circa 2009] SHARP filter:
16-bit undithered:

24-bit:

Clearly the TEAC has no problem with reproducing that least significant bit in the 16-bit signal. Also, obviously the resolution has improved significantly by going to 24-bits.

ASUS XONAR Essence One [2x BB PCM1795 c. 2009] (opamps upgraded to all LM4562):

Very nice... Notice a wee bit of channel imbalance - the left channel (blue) seems consistently louder than the right. Same internal DAC chip as the TEAC so similar level of performance expected.

Logitech Squeezebox Transporter [AKM4396 c. 2004]:

Nice! Not bad for a discontinued device from a computer peripheral manufacturer released in 2006, eh? ;-)
Of course, the Stereophile review demonstrated this nicely already...

Logitech Squeezebox Touch [AKM4420 c. 2007]:
WiFi (only 30% signal strength 2 floors up from router!):

Ethernet:

24-bit:

Three observations:
1. Clearly the Touch is noisier than the better DACs above. It's still capable of >16-bit dynamic range though.
2. Some DC shift is evident - look at the upward slope with the 24-bit sine wave and compare to the Transporter above. Maybe this could be improved with a better linear power supply than the stock switching wallwart I used... Not sure if an improvement would be audible however.
3. No substantial difference between WiFi and Ethernet. (No surprise; just thought I'd have a look to see if WiFi added much noise down at this level.)
N.B. Remember that this is still a pretty good result - we are looking at a waveform down at -90dBFS, or ~90 microvolts! Nice correlation with what Stereophile found (Figures 5 & 6) in terms of the Touch being a 'touch' more noisy than better DACs.

AUNE X1 Mark I [BB PCM1793 c. 2003] (using CM6631A USB-to-Coaxial S/PDIF, ASIO driver):

This is what can be achieved by a <$175 DAC off eBay direct from China these days (I bought this unit in early 2012). Notice that it's able to produce a cleaner analogue output than the Touch. But it's also not quite up to the standard of the TEAC, ASUS, or Transporter. Notice both a slight channel imbalance as well as mild amplitude fluctuations (again, possibly due to cheap wallwart). Hopefully the following zoomed out screenshots illustrates this well for comparison:

AUNE X1 - left channel (blue) noticeably louder and notice the amplitude fluctuations over time.

Touch - Notice it's more noisy with unpredictable amplitude spikes occurring in both right & left channels.

TEAC UD-501 - more stable, clean, uniform waveforms in comparison.

As much as it's great to see the level of performance afforded by DACs these days, I'm very impressed by the level of performance of this old E-MU ADC! As I have stated before, one of the reasons I put up these posts is to demonstrate that it doesn't take megabuck equipment to test out audio gear objectively. A lot can be "known" about the performance of a piece of hardware rather than depending on only subjective "opinion".

The other test I would categorize as the equivalent of "pixel peeping" is the Dunn jitter test where we're "peeping" into a small part of the audio spectrum around the 11 or 12kHz primary signal and scanning for sideband anomalies. IMO, neither the jitter nor this undithered LSB test really are that important for audio quality. Random noise affecting the 16-bit least significant bit would sound like some form of dither (eg. like what happens when an HDCD with embedded LSB data is decoded by a non-HDCD player). Likewise, my feeling is that even a "moderate" amount of S/PDIF jitter (like say 1ns) isn't going to intrude into my listening pleasure. (Maybe one could make the argument that the details and nuance of sound/music can reside in these microscopic domains but I have yet to see any proof...)

Assuming the digital player/DAC is meant to be faithful to the source signal and doesn't implement a DSP known to affect the LSB data, to be able to measure and verify precision down to these levels I believe would be a reasonable pre-requisite in achieving high-fidelity. It's a test of how well the hardware was designed and implemented than necessarily how it "sounds". Just like knowing if a hi-res digital camera is capable of the resolution it claims... You might never need 36 megapixels for a slideshow or in print, but it's good to know that the camera was capable of delivering on the claims! Likewise, if I'm going to spend a good amount of money on high-fidelity gear, I'd certainly like to know that precision engineering went into it by the results of tests like this among others already discussed over these months.

Let's throw some nostalgia in. Here's what the MUSE Mini TDA1543 x4 NOS DAC looks like down at -90dB (using the CM6631A USB-to-S/PDIF coaxial interface):

Party like it's 1991! Ugly... Clearly it's incapable of accurately reproducing the 16-bit LSB undithered tone.

Zoomed out (24/44) - still ugly:

Using a 24-bit signal makes no difference since this is a 16-bit DAC and the lower 8 bits get truncated. The Philips TDA1543 DAC chip was introduced back in 1991 according to the specs sheet... Thankfully, it looks like DAC designs have improved somewhat since then at least in this characterstic :-).

A stroll down memory lane... The top 3 highest grossing movies of 1991: Terminator 2, Robin Hood: Prince of Thieves, Disney's Beauty And The Beast. Top 3 songs (Billboard): (Everything I Do) I Do It For You, I Wanna Sex You Up, Gonna Make You Sweat (Everybody Dance Now). Hmmm... Good year :-).

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Folks, much of the measurements above were done about 3 weeks ago but I'm putting this post up behind the "Great Firewall" while on vacation (hey, >10hr plane flight gave me plenty of time to do some writing!). I know a few people are trying to get hold of me by E-mail. Unfortunately my VPN + Outlook is a bit finicky so other than more important work related matters as I check the E-mail every few days, I will likely not be responding until mid-August.

Time to go enjoy some good food... And snap some pictures of course... :-)

BTW: For those interested in some light non-fiction summer reading, consider picking up Chuck Klosterman's "I Wear The Black Hat: Grappling With Villains". An enjoyable, thought provoking social commentary.

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.

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) :-).

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 :-).

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.

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...

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"!

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!