Saturday, 5 April 2014

MEASUREMENTS: Nexus 7 to Audioengine D3 (A "Kinda Portable" Audiophile Playback)

Okay, for fun, I thought I'd grab a few measurements of something... Somewhat... "Portable" :-)



What you have here is my Nexus 7 tablet connected to an "on the go" cable (5" male microUSB to female standard USB, off eBay - pack of 3 for $10) --> Audioengine D3 DAC/amp --> Sennheiser HD800. Unfortunately, The D3 would not power up consistently when plugged into the Nexus 5 smartphone since that would have been even more portable! Looks like the D3 demanded more power than the Nexus 5's USB port could deliver.

I got USB Audio Player Pro software for the Android in order to get the USB DAC working. Unfortunately USB Audio Class 1/2 devices are not supported by Android by default. You can see the basic interface above (I happened to be playing some Bruno Mars Unorthodox Jukebox). It recognized the D3 without a hitch. It sounds the same to me connected to the HD800 like the other machines I tested the Audioengine D3 with last time so no need going into any subjective evaluation here.

I was more interested in whether the objective data showed any difference between this "mobile" set-up compared to the laptops / desktops.

RightMark Results:

Remember the clipping at 100% with the Audioengine D3. All the measurements are done with hardware volume attenuated to 92%.

16/44:

Frequency Response
Noise Level
THD
IMD
24/96:

Frequency Response
Noise Level
THD
IMD
The first column in the summary tables is the Nexus 7 + Audioengine D3. The second column is the ASUS Taichi laptop connected to the Audioengine D3. Finally the 3rd column is the Nexus 7 natively without using the external USB DAC.

As you can see, there is no substantial difference whether the ASUS laptop/ultrabook or Nexus 7 was used in either the 16/44 or 24/96 test case. The DAC determines the final audio output, not the source "transport" device.

The Audioengine D3 is substantially better as a DAC of course. As you can see, the Nexus 7 + D3 easily outclasses the native Nexus 7 audio output off the headphone jack. It has a flatter frequency response with lower noise floor measurable even at 16/44. The difference is more evident with a 24-bit audio signal... The 24/96 frequency response demonstrates that the native Nexus 7 in fact is incapable of 96kHz sample rate.

Jitter:



Slight difference between the 2 J-Test measurements. The noise floor seems a little bit higher on the whole with the Nexus 7 when I ran this test making the peaks such as the 16-bit modulation pattern less obvious. Also a little cleaner around the 24-bit 12kHz primary tone with the Nexus 7. The differences are down around the -120dB level and would not be audible IMO.

Conclusion:

Okay... So this set-up isn't exactly a portable device. But it does demonstrate that you can get excellent sound out of a small Android device with the USB DAC that is objectively equivalent to using a standard computer.

Truth be told, I don't need "audiophile quality" sound in a portable device. I can't remember the last time I listened to my smartphone or iPod somewhere quiet with expensive headphones. On-the-go, convenience trumps everything else IMO so there'd be no way I'd bother with full sized headphones. If I did bring full-sized cans around on a train/plane/subway, it certainly would not be the expensive high-resolution open design headphones!

Day to day, I have my Nexus 5 phone with me. There are a few albums saved on the phone as MP3 or FLAC when I want to listen. Otherwise there are countless apps to listen to Internet radio and music streaming services which is what I listen to most. The days of the isolated non-networked portable music player are long over for me and probably the vast majority of music lovers.

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For fun I decided to jump on the Geek Out bandwagon for a spin and see how it goes.

Looks like we're starting to see some user reviews coming out now, the first more formal one I see being this one at Part-Time Audiophile. So far it looks encouraging. As expected there are a couple of comments about the heat production of the 1W model. No comment about how it handles DSD and it looks like those guys are Mac-centric, so not clear how it works out in the Windows world (ie. drivers). For what it is, I find purely subjective reviews interesting to read but I would prefer something a little more than the usual "drive by shooting" ;-).

After a bit of humming and hawing, I decided to go for a blue "Super Geek" ($250, 720mW, 3.4Vrms peak) model as the best compromise for my case. My rationale is simple... The amplitude difference between 720mW to 1W is only 1.4dB (as compared to 2dB between 450mW to 720mW). I'm still concerned about heat production since this is a class A design which sucks full power all the time... Assuming the enclosure design for heat dissipation is the same, the lower power model should drop the running temperature a few degrees. I can see myself using this as a line-level DAC if it measures really well and tap the headphone amp feature only on occasion (the Audioengine D3 appears lighter and smaller for travel). As a 'standard' USB DAC, this also means it'll likely be "on" all the time so I'd rather not have something too warm sitting on my table. Furthermore, with a laptop computer, an inefficient class A device means more power drainage. The $50 saved is trivial for this hobby and not really an issue.

Anyhow, once I get this, I'll let you know some results... It will be interesting to see how this compares to the TEAC UD-501 which has the same DAC chip and similar feature set (NOS, DXD, DSD, etc...). Hopefully it doesn't take too long to ship out - my understanding is that only the 1000mW "Super-Duper Geek" model is released so far.

Juergen mentioned having a look at the HpW-Works software package for jitter analysis. Might just do that although I remain unconvinced it makes any audible difference in 2014 especially with asynchronous USB. I have yet to see a good example of a decent modern piece of equipment where jitter can be shown as the culprit for impairing the sound quality.

Tonight's music:
Kodo - Mondo Head - Although I generally prefer the more traditional sound of Tsutsumi, this one not only sounds nice in stereo but fantastic in multichannel off the SACD!

Enjoy the tunes everyone!


22 comments:

  1. Resonessence Herus – Jitter

    Archimago: I really would give HpW-Works a try. With 16 Bit Data, nowadays all asynchronous DACs behave very well, also with 1FS (44k1 and 48k) 24 Bit Data. But when you measure 2FS (88k2 and 96k) with 24 Bit, you will clearly see a lot of differences. Also your noise flour and shape of the 24 Bit Jitter Test is not looking “perfect”. I personally have / had also the EMU 1616M, 1212M, 0404 USB and 0202 Tracker and they were very good for the price for using as test ADCs, so I know, it can be better with your 0404 USB Interface as test ADC.

    If you want something better, but still very reasonable in pricing, you can get the RME BabyFace Interface and this one, besides beeing more “modern” and still in production and supported, shows even slidly better measurements as a ADC, than the EMU 1616M. It have also fully balanced analog in and can measure 192k. But as mentioned above, give HpW-Works a try with 88k2 / 96k and 24 Bit Jitter and you will see differences (for sure).

    Resonessence Herus: As for portable on the go, I have this small USB DAC. It has a very clever start up EPROM, so that is works even with iPhone (from iOS7 up) and iPad, both with CCK (camera connection kit). As a player on my iPhone, I have the Onkyo HF Player, and this player supports all four lossless files (Wave, AIFF, FLAC and ALAC) with upt to 24 Bit and 192k and also native DSD via DoP. You should give this one a closer look. I do not know, what other portable USB DACs will boot up on portable devices (and the Herus does support DXD and 2xDSD).

    Good luck
    Juergen

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    1. Thanks for the tips Juergen. Interesting and worth looking into...

      Here's the thing... Do you think jitter is audible from your tests even though measurable - for example with the differences at 2fs?

      Indeed the jitter test at 24-bits with a 12kHz signal isn't perfect but at what point is perfection unnecessary and have we already surpassed the "good enough" point in all this? For example, measuring 2fs jitter; is that like scaling up the J-Test here - measuring the accuracy of a 24kHz primary signal in 24-bit with a low level 500Hz jitter modulation tone at 96kHz? I suspect indeed that will show significant differences between DACs as you suggest since timing difference/imprecision will be magnified, but how relevant would that be to human hearing?

      I can see in some cases this would be very useful of course... For example, if I were doing the hardware engineering with state-of-the-art DACs and want my next DAC to be even more precise. Just not sure how important this would be for the home listener.

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    2. The jitter source have to be in 16bit. While its a clear sine (coherent) SR / 4 and an LSB toggle. Interesting is on the sample signal:

      - The symmetric side band freq. around the source sample rate fs /4
      - How side band noise on source fs / 4 behaves (high or low side noise)
      - How the low level LSB toggle (a none symmetric DC signal) in Fourier terms behave

      Hp. Widmer (HpW-Works)

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    3. Different J-Test Signals

      Dear Hans-Peter Widmer. I have your HpW-Works for some years now, because it is a good tool when I am on the road (just with a portable AD/DA Interface), to test some audio devices (also together with RMAA).

      Your jitter measurements is working very well and does correlate to that of my Audio Precision system, except one difference. Your test signal is a bit more “relaxed” than that of Audio Precision (Julian Dunn).

      The difference lies in the phase shift, where the sample points are take. Both, your and AP test signal has for ¼ FS signal around – 6 dBFS, but you have taken that at the peak of the signal and AP at 90 degree phase shifted.

      This ends up that even you both have the same sample value, that the maximum “intersample” or analog value (after DA and reconstruction filter) of the AP system is about 3 dB higher than yours. So the measurement of the AP system looks a bit worse than of yours.

      Just for your to know. I do not know what version of the signal is right or wrong (yours looks more right than that of AP, but know that Julian Dunn stood behind the jitter test at AP makes me thinking that maybe the “stronger” signal is the “correct” one).

      I would have screen shots of both versions of the signal, but I do not know how to add this in blog.

      Best Regards
      Juergen

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    4. Hi Juergen,
      Yes the older V2.x SW release do not have the D-Jitter signal implemented. The new V3.x release has in D-Jitter signal included.

      Also, I got some feedback using my alike D-Jitter produced some higher side band distortions..

      I love to see your measurements, if possible share it using my support @ hpw-works (dot) com

      Thank you

      Hanspeter

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  2. Archimago

    In the last 3 months, I have done a lot of investigations into Jitter, but all with the same DAC chip, same power supply, same output filter, so I have had always the same THD, Noise, Frequency, … and only the Jitter as variable point.

    As for 1FS (CD Audio Data), with a regular good asynchronous USB interface, you are good enough, that you do not see (measurements) or hear any differences, but with 2FS 24 Bit Data, there is clearly an audible (and measurement) difference.

    The stability of the soundstage and the air between the musicians is different, with different jitter levels. When I was able to improve the jitter measurement with the 24 Bit 88k2 data, I could hear also a difference with 24 Bit 88k2 music, but I couldn’t with CD Audio data.

    I hope it is clear what I mean, I have had everything the same except the parts, they determine the jitter. As I mentioned also in a different reply, I have (as long time experienced audio engineer) also an Audio Precision measurement set, but with HpW-Works software and EMU 1616M hardware (or the RME Babyface as ADC), I could have a very high degree of measurement correlation.

    Good luck (and it is worth also to look at the very clever boot up sequence of the Resonessence Herus, that allowed the direct connection to portable iOS or Android devices and playing back also High Res files natively (but yes, with a sort of limitation because of only limited power supply voltage and current and also limited space of parts).

    Juergen

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    1. Some questions regarding the listening tests performed here, as I like to understand more about the test conditions.
      Note that I am not questioning your listening skills/experience/expertise here but genuinely interested in the subject.

      At the moment of the listening test did you KNOW which file was playing or were you able to confirm the audible differences by truly blind tests with at least 10 sessions (changing only 1 parameter) and statistic relevance ? Did you keep records ?

      Has the DAC used a (NOS) ladder type chip or Delta Sigma chip ?

      Also for the 44 and 88kHz was the same 88kHz file used (downsampled with a known well performing algorithm ?) or did you use a 44kHz file which was upsampled to 88kHz (so that there is no difference in audible and ultrasonic contents) ?

      How did you improve the jitter (and thus audible qualities) ?

      Did the same phenomenon occur with different DAC's as well ?

      When you mention:
      - The stability of the soundstage and the air between the musicians is different -
      Would you think that 'soundstage' and 'air' is determined by timing issues OR by LEVEL differences ?
      AFAIK, ultimately, in a DAC all timing differences are converted into (extremely small) level differences by the post filtering.

      Do you think it might it be possible that 'soundstage' and 'air' are located in that -100 dB (considered inaudible by some) range and can affect perceiving sounds differently at a subconscious level, but still clearly audible enough if one looked for these specific properties (air, soundstage) in a recording ?

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    2. Some background information

      First of all, no, I do not have done a double blind test, this was not possible in reality, because I was aware which input receiver PLL unit I plugged into the main dac board, I used a multi level delta sigma DAC I am familiar with for nearly eight years (Crystal CD4398).

      No, for that test, I use different files for 44k1 / 16 Bit as for 88k2 / 24 Bit. Herewith my target was not to compare different file formats. For listening in 44k1 / 16, I have the same 20 files for years, so I am used to it, the same belongs to the 88k2 files.

      As for soundstage and air, I have to add, that besides being an audio engineer, I am also a hobby musician for more than 35 years (still practicing) and also a part time recording engineer (have a small recording studio) but doing mostly classical recordings in concert halls.

      Therefore, I use mainly only 2 small membrane condenser mics in ORTF / OSS configuration, recording in 882 or 176k2, without EQ or Limiter or Compressors, just plain 1:1. Being there during the recordings, I am aware, of the real soundstage and musicians placements.

      As for jitter reduction, I have tested different input receivers, different PLL settings, analog versus digital PLL, combinations of analog and digital PLL, VCXOs, with and without FIFO.

      But as mentioned above, I do not want to teach about what jitter level is audible or not, I just want to recommend a slightly better J-Test program, in order to get more reliable data.

      What jitter numbers are audible and what are not is not my discussion here (because this will be endless), I just want to help to “improve” the measurement on this interesting (and different) site.

      Juergen

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  3. PS: J-Test

    Archimago. I originally started my suggestion of using HpW-Works for Jitter measurements, because I mentioned already, that your J-Test graphs are not looking „right“ / correct.

    The RMAA measurements are fully correct and this is a very handy program for „DIY“ tester, but it would be nice, if the j-test graphs would have the same „quality“ to judge the results.

    And then you will notice that the Teac UD-501 DAC does have some “problems” with 24 Bit J-Test signals. The slightly above Mytek 192 DSD DAC is better in that regard and the Benchmark DAC 2 HGC is very good in this field. This is btw, a very good sounding headphone amp. (I have / had them all for comparison and have measurements (and listened to) of them all).

    Juergen

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    1. Thanks for the detailed response Juergen. Interesting... I'll see what I can do about learning to use the HpW-Works package... May have a few questions for you on that in the days ahead :-).

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  4. Hi,

    Have you considered measuring an Ipad with a cck into a D3 or USB DACs like HRT Microstreamer ? Curious to know whether it measures as well or better than a Nexus tab.

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    1. I do have an iPad and CCK so can have a look at this time permitting... Wondering if you know whether special drivers or player programs are needed with the iPad/iPhones?

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    2. Ipad CCK

      Three points on that.

      Software: With the iOS Music Player, you can only play and sync with iTunes up to 48kHz 24 Bit max. But when using an external USB DAC via CCK you can go up to 192kHz 24 Bit PCM with FLAC Player (FLAC files only) and even to DSD64 with Onkyo HF Player (AIFF, Wave, ALAC, FLAC, DSD), and with both, the audio data are not synced via iTunes music library, they are feed through the embedded files storage function.

      Hardware: The CCK does only work with USB DACs that need low current from the hardware side and does also report back on the handshake protocol, that they need low current. With USB Audio Class 1 USB, this was easy to achieve, but with USB Audio Class 2, most of them need too much current when they are USB V-Bus supplied only (as with portable USB devices without batterie are always). You can not install any driver, the connected USB hardware has to fulfill this restrictions.

      As I mentioned before, the Resonessence Herus does support a clever boot up routine, so is supported from the iOS side to be connected via CCK and can play back High Res with low R headphone out and a clever software synced hardware volume control, because otherwise DSD playback can't be volume controlled via software volume control natively. Very clever design.

      Juergen

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    3. Hmmm, just tried the CCK with an old iPhone 4 and iPad (1st generation). Both machines complained of not having enough power to run the Audioengine D3.

      Perhaps the newer Apple machines would have more power for this USB DAC...

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    4. For the older 30 Pin CCK it is useful (not necessary, but it would be better) to connect the USB cable from the USB DAC into the CCK first, and after that, to the iOS device. This order is not necessary for the lightning connector CCK. Then the USB DAC will work, but only, when it does fulfill the needs of low current use and also declaring low power use in handshake protocol.
      Juergen

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    5. Thanks JR. Gave it a try with CCK + DAC then to iOS device... Still no go with the D3. Guess current requirements just not low enough.

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    6. As I mentioned above, most USB Audio Class 2 Devices need too much current for iOS Devices. But the Resonessence Herus does (due to a clever start up).
      Juergen

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    7. Discrepancies in Audioengine data sheet

      At their homepage, I can find, that the D3 does accept 192 kHz (Input Data Rate) and that the DAC does handle 192 kHz, but when using the TAS1020B USB chip, this is definitely a USB Audio Class 1 chip, that can handle 96 kHz max and the HD indication does also indicate 96 kHz max. So just ignore my writing about USB Audio Class 2 devices and the higher supply current.

      Juergen

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    8. yeah, I noticed that as well... The AKM DAC can do 24/192; I presume that was what they were referring to.

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  5. Even the HRT Microstreamer can achieve this as it needs a low current (100 mA). There is a workaround - using a powered USB hub. Again, it will be good to see if this kind of setup can match a laptop in terms of measurements.

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    1. Yes, I know, powered USB hub do circumnavigate the low current requirement between iOS devices and the input of USB DACs but this is not a real practicable solution for on the go.
      Juergen

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  6. I just wanted to say...thanks. I've been a big audio freak for years (used to sell audio and computer gear in the early 90s, and then moved on to working for Kenwood. I'm a software engineer now, but I'm also an electronic musician and producer. I've always known MP3 files were a bad compromise, particularly when I fell for AAC and WMA files years ago, but the problem didn't really "blow up" until I started rendering my 24-bit/48 kHz files from various DAWs and converted them via LAME, and also uploaded them to SoundCloud for transcoding and had all kinds of bizarre muddiness in the mid-bass (the bass was shifted upward, at least from what the waveform seemed to indicate), and I found I had to back off my mixes a bit when rendering to 16-bit/44.1 kHz to avoid ISC (Inter-Sample Clipping) issues once going to MP3. Using iZotope Ozone 5 and a few other tricks, I seem to be able to get WAV, AIF, and MP3 files I can accept. However, the MP3 will still have subtle differences in most of the audible spectrum, and really odd artifacts when strong china-cymbals or things with strong, light or uncompressed transients come up.

    I'm now re-ripping my CDs and went to Beatport to upgrade my non-WAV downloads to AIFF so I could "start over," and things are a lot better. I use a FiiO E18 Kunlun DAC/Amp with my Galaxy Note 3, but I'll probably move to a different portable DAP (like the X5, or even the A&K 240) to have enough storage. I typically have around 16 GB of my existing collection in MP3 with my phone, so I'm probably going to need a good bit more storage and have enough room for a few videos. :-)

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