If you look back at my writings and tests over the last few years, you will see a number of instances where I did measurements using different operating systems, computer hardware, different bitperfect software (for Windows, for Mac, "audiophile" JPLAY), and looked at things like computer CPU loads and jitter. Already, in the post on the AudioEngine D3 review, I demonstrated that even with disparate computer hardware and operating systems like Windows 8.1, Windows Server 2012 R2, Mac OS X "Mountain Lion", and "Mavericks", I was not able to show measurable differences in noise level, dynamic range, distortion or jitter with this DAC which operates up to 24/96.
The results are not surprising really, as a computer audiophile in 2015, it's quite likely that one would be using a high-quality USB DAC which is asynchronous, thus avoiding issues like the potential for significant jitter due to timing problems in the interface (more below). Yet, I get questions still about uncertainty as to whether different OSes make a difference and then there are programs out there like Fidelizer (check out this review, or this one) and even the well popularized Audiophile Optimizer (through reviews like this one) claiming improvements; strangely none of those reviews bothered to tell us exactly what DAC was used!
Could the OS and "optimizer" programs like these affect the output through the OS audio stack (eg. Windows Mixer) or the motherboard headphone jack? Sure... I guess it's quite possible since process scheduling, mixing and dithering algorithms could change. But seriously, what audiophile interested in high fidelity output with dollars invested in good digital source/pre-amp/amp/speakers/headphones would be listening to motherboard audio or DirectSound with Windows audio? Not many I bet. As usual, despite the testimony from various places about how Windows 10 has improved sound, I believe we have yet to see any evidence that sonic output of a decent modern asynchronous USB DAC using a bitperfect driver like ASIO or WASAPI has changed with an OS upgrade (or the use of OS optimizations).
First, let's talk about Windows 10. Not only is this the "latest and greatest" Windows, there is the potential that this is "the last" Windows version for the foreseeable future. Maybe the situation will be similar to "OS X" in the Mac world where that moniker has stuck since 2001. Windows 10 does add a very useful feature - multiple desktops - which I use regularly now on my main workstation. The "Start Menu" is back rather than using that un-desktop-friendly "Start Screen"; makes locating apps and installed programs much easier. Beyond that, it really hasn't changed the way I use the machines; I have not tried Cortana and the new Edge browser shows promise but is not up to the features of Firefox or Chrome yet.
Since I'm actually doing the upgrade, what better time than now to measure whether Windows 10 will change the sound of my DAC (the TEAC UD-501) connected to the HTPC machine before and after the upgrade?
I. The Hardware & Software
The test computer is the following (similar to the previous HTPC build article with slight changes):Enclosure: Bitfenix Prodigy M microATX
Power supply: fanless SeaSonic SS-400FL2 400W
Motherboard: ASUS B85M-E/CSM - has HDMI with limited 4K (30Hz) capability
CPU: Intel Pentium G3220 (dual core, 3GHz, 54W TDP only
Haswell graphics features but slow 3D)
- underclocked to 2.8GHz, slight undervolt
CPU Cooler: CoolerMaster Hyper 212 Plus (total overkill but running fanless!)
SSD: SanDisk Ultra II 240GB SSD
RAM: 8GB Kingston DDR3 1600
(note the G3220 CPU will underclock this to 1333)
Normally I use the Corning Optical USB3 cable with this computer, the rationale as previously reported. In order to remove another variable, for these tests, I switched back to a generic good quality 12' shielded USB cable.
The humble Pentium fanless HTPC. Generic (silver) USB2 cable shown. |
Windows 8.1 Pro (latest updates as of August 10, 2015) will be upgraded to Windows 10 Pro (with all updates as of August 11, 2015).
A couple of other tweaks thrown in the Windows 10 tests to explore software optimizations:
- Fidelizer 6.8 Demo + foobar
- Fidelizer + JPLAY 6.2 Demo using foobar
I trust these combinations will demonstrate enough about whether the Windows 8.1 to 10 upgrade changes the sound of a typical modern USB DAC. Furthermore, let's see if OS optimization with Fidelizer makes a difference, and now that it has been 2 years since I last looked at JPLAY, whether anything has changed... Note that because there are many potential settings in programs like JPLAY and Fidelizer (which of course appeals to the tweaking demographic), I'll measure a typical recommended setting to see if there is any evidence of a difference.
Set-up is as follows:
HTPC (Win 8.1/10) --> shielded 12' USB cable --> TEAC UD-501 DAC --> shielded RCA --> E-MU 0404USB ADC --> shielded 6' USB --> Win 8.1 measurement laptopAll test audio files situated on the computer's SSD except where indicated for "streaming" off the ethernet from my Windows Server 2012 R2 machine situated about 50 feet away (probably ~70' ethernet cabling snaking through the walls and corners) in an adjoining room with 2 inexpensive gigabit switches in between (D-Link DGS-1016A located by the ethernet central patch and TP-Link TL-SG1008D in the soundroom about 12' from the computer). All tests performed with bitperfect ASIO drivers for both playback and recording unless otherwise indicated.
System running DMAC test (see below) - Win 8.1 still - note TV screen with Win 10 advert bottom right. |
And we're off with the upgrade... |
II. Windows 8.1 vs. Windows 10
First let's start with just a look at the OS change itself, as usual, here's the RightMark 6.4.1 Pro summary; I will just measure "high-resolution" 24/96 performance since 16/44 is no challenge for most DACs these days:No difference at all...
Frequency Response |
Noise Level |
IMD+N |
Any evidence then of jitter differences?
Slight variation at the level of the noise floor. Nothing of significance IMO. As expected with the low noise TEAC DAC, it's very easy to see the jitter modulation LSB component in the 16-bit signal.
III. Let's add Fidelizer 6.8 Demo
So far, nothing exciting between Windows 8.1 and 10. What about an OS "optimizer"? Well, I followed the Fidelizer instructions to change some foobar settings as recommended including: driver as WASAPI (event), 50ms buffer, full file buffering, high worker priority, use MMCSS, MMCSS mode "Audio". Then I installed Fidelizer in "Extremist" mode, telling it to use foobar...
Here's evidence that it does do a few things to the OS like turning down the number of services/processes/threads:
59 --> 51 processes. And 893 --> 705 threads after "Fidelization"... Ok, cool... |
And does this affect the asynchronous USB DAC's output?
Frequency Response |
Noise Level |
IMD+N |
Clearly the answer is that nothing of significance has changed.
And for jitter?
Dunn J-Test spectra again essentially the same as above... A little bit of variation at the noise floor below -120dB.
Measurements now again using foobar but with the JPLAY ASIO output "device"...
Nada... No difference!
And jitter?
Oh boy, we have a problem... Recall that in my testing back in June 2013, I said JPLAY seems to have a problem with Kernel Streaming 24/48? It doesn't appear to be handling the 24-bits properly and instead we end up with around 17-bit resolution at best... Well, guess what, I suppose nobody told the developers about this issue! I double checked again and it looks like if I switch to JPLAY with the ASIO driver this problem goes away. Here are the measurements with the "Classic" Engine using ASIO compared to "ULTRAstream" Engine with Kernel Streaming:
Notice the worsened noise level with Kernel Streaming compared to ASIO. And here is the noise profile:
Considering we have gone from version 5.1 in June 2013 to 6.2 today, let's just say it's a bit "disappointing" that 24/48 is still not being managed correctly by a piece of software billed as "One player to rule them all: enhance sound quality of your player"! I fail to see how any of this can be construed as "enhancing" sound quality. And I'm of course sure many "golden ears" have been deprived of hi-res 48kHz audio enjoyment because of this bug over the last few years :-).
[BTW, my guess is that JPLAY flipped the "endianness" of the lowest 8 bits in a 24-bit sample to cause that anomaly on playback.]
IV. Let's Add JPLAY 6.2 Trial on top of Fidelizer and Windows 10!
Okay, let's now get truly "audiophile"! What if we installed JPLAY ("Optimize JPLAY for single PC setup" checked during installation) and tried it with some pretty "strong" settings with Windows 10 and Fidelizer activated?Kernel Streaming, "ULTRAstream" engine whatever that means, and DAC Link down to "~2.5Hz" from the default of "~10Hz"... Sounds fancy... |
Frequency Response |
Noise Level |
IMD+N |
And jitter?
Oh boy, we have a problem... Recall that in my testing back in June 2013, I said JPLAY seems to have a problem with Kernel Streaming 24/48? It doesn't appear to be handling the 24-bits properly and instead we end up with around 17-bit resolution at best... Well, guess what, I suppose nobody told the developers about this issue! I double checked again and it looks like if I switch to JPLAY with the ASIO driver this problem goes away. Here are the measurements with the "Classic" Engine using ASIO compared to "ULTRAstream" Engine with Kernel Streaming:
Notice the worsened noise level with Kernel Streaming compared to ASIO. And here is the noise profile:
Considering we have gone from version 5.1 in June 2013 to 6.2 today, let's just say it's a bit "disappointing" that 24/48 is still not being managed correctly by a piece of software billed as "One player to rule them all: enhance sound quality of your player"! I fail to see how any of this can be construed as "enhancing" sound quality. And I'm of course sure many "golden ears" have been deprived of hi-res 48kHz audio enjoyment because of this bug over the last few years :-).
[BTW, my guess is that JPLAY flipped the "endianness" of the lowest 8 bits in a 24-bit sample to cause that anomaly on playback.]
V. Music Samples with Audio DiffMaker
Up to this point, the results are from synthetic test tones meant to explore parameters of resolution of the audio output. One criticism I've heard about objective testing in general is that "nobody listens to test tones!" To address this concern then, I will bring back the Audio DiffMaker Protocol that I put together 2 years ago. The test consists of segments of 4 actual pieces of music spliced together as a composite lasting ~30 seconds (which I call the DMAC - "DiffMaker Audio Composite"). I played the DMAC audio file with each of the conditions through the DAC and record them with the ADC. The computer then calculates based on what was "heard" the "null depth", a value of how much similarity there is between the "reference" and "comparator" recordings. Any change - frequency response changes, distortion, noise, severe timing issue - will be accounted for in this test! The greater the null depth, the "more alike" the two sound. Realize that with analogue output, it's impossible to be 100% the same... For example, as the DAC and ADC heat up, small variations happen including "drift" in the samplerate, this is why many of the high-end DACs these days advertise TCXO (Temperature Controlled Crystal Oscillators) to reduce the phenomenon. (While technically this is good especially in studio settings where synchronization is important, the actual value in terms of audibility in home playback I suspect is questionable.)Using my procedure and the E-MU 0404USB ADC, I can reliably achieve null depths of around 80-90dB with good inter-test reliability (average 80.2dB, standard deviation 3.9 based on previous work) for bitperfect lossless audio. Much of this is dependent on the reliability of the DAC and ADC of course in their ability to play back and record with high fidelity. With 320kbps MP3 (well encoded with LAME 3.99), null depth drops to ~65dB, telling me that the measuring device can "hear" approximately 15-20dB of difference between the lossy compressed audio versus the original lossless source averaged over 30 seconds. Considering that humans rarely are capable of differentiating 320kbps MP3 consistently (and usually only with specific segments of difficult to encode audio samples) and echoic memory lasts <5 seconds, digital differencing test paradigms like this are IMO certainly better than the best human ability to recognize a difference between music samples.
Let's see then whether the machine can hear a difference between the different OS's, "optimizations", and playback conditions. All recordings were done at 24/96 (original lossless DMAC audio is 24/44). The "reference" recording is with Windows 8.1 using foobar, ASIO driver, default settings. For each condition, I measured 3 times (except for the MP3 test conditions measured only 2 times as sensitivity testing). Each data point you see is the average of the two channels' calculated null depths.
As you can see there are quite a number of conditions shown in the graph. "Win 8.1" and "Win 10" are just the standard playback results using foobar, ASIO driver, default settings off the local SSD storage. The "Stream" conditions are when I play the audio off the network through my Windows Server 2012 R2 machine 50' away over gigabit ethernet (just standard SMB file sharing). The "MP3" conditions are for "sensitivity testing" to see if MP3 "sounds" consistently of lower accuracy to the machine (they obviously do!).
The bottom line is that Win 8.1, Win 10, ethernet playback, Fidelizer, and JPLAY made no difference as far as I can tell in terms of what is likely to be audible differences. The variation is well within inter-test differences and generally stays around 80dB null depth over the 30 seconds of the music composite for every condition except with 320kbps MP3 (where it sits around the expected 65dB area). Notice that the chart is arranged in temporal sequence; that is the Win 8.1 tests were done first, then Win 10, and the last to be done were the Fidelizer and JPLAY tests. Over time, as the equipment warmed up the drift appears worse but settles down. The biggest change was between Win 8.1 to Win 10 during which time I let the machine spend a few hours downloading the OS, installing, etc. before I came back to run the tests. In retrospect, I probably should have given the system at least a couple hours to warm up (instead of measuring in the 1st hour) - might have made the data a little more consistent comparing the Windows 8.1 results with the others but conclusion would be the same.
VI. Conclusions
I hope these experiments and results add to the discussion around operating systems and sound quality using a "modern" DAC and standard computer source as a case study.The bottom line is that I am unable to find evidence to say that OS changes will make any difference to analogue audio output from a modern asynchronous USB DAC despite having heard comments already that somehow upgrading to Windows 10 will make things sound "better". Likewise, I see no evidence that software like Fidelizer will do anything to the sound output even though I can show that it "worked" to reduce the number of processes and threads of execution in the OS. Finally, and yet again, JPLAY demonstrates no clear ability to affect the sonic output and in fact (again after 2 years since the last test) I see problems with Kernel Streaming in playing back 24/48 accurately. This is obviously worse than "no effect" in that it actually deteriorates the quality of playback. I seriously wonder how these developers actually perform testing on their software and what they're actually doing! (Beyond marketing and dreaming up terminology like "ULTRAstream" for example.)
I see that Audiophile Optimizer has a bit of a following but given that there is no free trial link available for download (you can contact them for some kind of trial version / 14-day policy). I certainly encourage others to perform an objective assessment of whether claims about Windows Server 2012 R2 and software optimization can make a difference (already with the AudioEngine D3 tests last year, I found nothing special with Server 2012). As usual, the developer does not provide any objective evidence except testimony. Given the results here, I think it's fair to remain skeptical of claims and would strongly suggest some kind of "try before you buy".
As I mentioned in the intro, Windows 10 itself brings us back to a more useful desktop computing experience. So far I like it. I have thus far upgraded 4 machines - 2 for work, the HTPC and my main 4K home workstation. For the most part, upgrades went smoothly. Ironically the most trouble I have had so far is with Microsoft's own Surface 3 Pro! I kept getting stuck at 18% install and discovered it had to do with MBR (Master Boot Record) issues. I found this link and the instructions by funnyfarm299 to do the "bootrec /rebuildbcd" before it finally installed. I actually still have some issues with the Surface and WiFi connecting sporadically and at times getting kicked off the network - I hope MS fixes this ASAP. I think this has something to do with VPN software like Cisco AnyConnect in some situations. The only other issue I had was sometimes a couple drivers and programs did not migrate properly. For example I had to manually direct Windows 10 to look in the "C:\Program Files\TEAC" directory to find the 64-bit DAC drivers after the OS update. JRiver 20 also needed a reinstall.
Anyhow, enjoy Windows 10 if you decide to install. "Win X" could stay with us awhile! And that's probably not a bad thing... I have been using Windows 10 with my other computers over the last few weeks and have not noticed any sonic difference to speak of among the various machines and different DACs (like the ASUS Essence One listening with headphones).
Disagree with my conclusions? Please experiment and if there's data to suggest I am in error, please leave me a comment and links! Thanks.
----------------------------
Since I will be away on vacation for a few weeks, I want to end off with a couple of "musings" and comments I have "heard" on line.
1. Let's talk about the phrase "bits is bits". If it is true that "bits are NOT bits", then what is it that is being transmitted? The standard answer is: bits (data) + timing. This is the reason why jitter has played such a big part in digital audiophile discussions. Jitter being the slight timing inaccuracies in the electrical signal which may affect the arrival time of the data (usually from the computer/transport to the DAC). And yes, I agree - so maybe that means I'm not a "bits is bits" kind of guy after all?
However, technology is evolving and improving. And while timing is a factor, the amount of timing irregularity in a typical USB data transfer has limited effect on good modern DACs. It is because asynchronous protocols have decoupled the need for accurate timing to the point where to all practical purposes, precise timing of data transfer from computer to DAC is just no longer a significant contributor to the final analogue output quality (we're not talking extreme data under-run timing issues of course). An asynchronous USB DAC can tell the computer to start and stop transmission based on its needs to fill the buffer. The buffer data is then fed to the DA converters synchronized with the precise internal clock (not USB clock). This is a significant advance over the "old" days with SPDIF clock recovery (see what happens with TosLink loopback here) and isochronous USB (see further tech details here); it was not difficult to show these timing anomalies with the Dunn J-Test, but these days, decent asynch USB DACs are essentially immune. Jitter anomalies then become a function of the DAC hardware itself. If we see bad jitter with an asynchronous USB DAC, the problem resides in the DAC and unless we're dealing with driver/firmware bugs, it's time to buy better hardware... Forget fancy cables making a difference, and likewise OS upgrades, optimizations, and "audiophile" software are unlikely to "fix" anything. In fact, I would say that installing these "tweaks" are more likely to cause functional issues or introduce bugs as we have seen.
I would not be surprised to see in the days ahead a significant dissipation in discussion around jitter and more focus on parameters like noise and power-related phenomena. Not that they're problems necessarily in most circumstances, just that worries about audible jitter have run their course...
Now if the "bits are NOT bits" folks have any other theories than the one above, I welcome the opportunity to "hear" about it and even better, providing evidence would open up dialogue and improve our overall understanding.
2. I believe I have been accused of trying to "prove everything sounds the same" by some (subjectivists) these last few years. Recognize that many of my posts have shown changes and difference; from jitter anomalies, questionable performance in certain DAC/ADCs (eg. the Dragonfly, Tascam UH-7000), to room measurements... Sometimes very much unexpectedly (like the JPLAY KS 24/48 issue above).
Sure, I'm trying to prove/disprove various ideas/claims but mainly for myself out of curiosity so that I may understand and explain when asked. I write because I've enjoyed this hobby for decades and in the pursuit of "high fidelity", had reached a point in this hobby where intellectual consistency and (I believe) honesty compelled me to experiment and explore beliefs which can be empirically verified. (Yes, I've been there over the years with claims like green CD rim pens, Belt Rainbow Foils, freezing CD's, etc...) I remember talking to other audiophiles over the years and there were so many questions, yet so few seemed to have any evidence to support what on many occasions appear to be fanciful beliefs! In the process of writing, I have thought about the philosophical "boundaries" between the subjective and objective. The differentiation of statements and beliefs born of the "magisterium" of art (impressions, feelings, aesthetics, preference) and that which belongs to science (design based on engineering principles, electrical energy, and sound waves governed by laws of physics). Both having value, but not always concurrently applicable to the question at hand and with different levels of generalizability. It has certainly been enjoyable sharing these findings and ideas.
My hope has always been to demonstrate that even a "guy in his basement" without the need of thousands of dollars in audio analysers can figure out for himself what is fact and what is hype. (Of course a big thanks goes to all the guys/gals for the discussions, tips and ideas to try along the way!) The fact that I cannot find differences some of the time is not because I'm "trying" to prove that none exist. Surely one must accept the possibility that there perhaps never was a difference to begin with given the many dubious claims! Objective results may not correlate with some things like level of enjoyment or personal satisfaction, but it can show what is factual, the magnitude of an effect, and fidelity to an ideal. When I put money down to buy audio products, I do so with the expectation that claims made correlate to real benefits based on applied science (quality engineering); I don't know if anyone would openly accept that they buy audio gear knowing that they're after a placebo, spiritual, or magical outcome. In writing about my experiments in audio, I hope that this adds to open discussion and raise the level of discourse and knowledge among audiophiles. It is a shame that the "professional" audiophile press and spokespersons do not seem interested in answering basic questions like the topic of this post nor in dispelling myths... Sadly, at times they even seem to be actively promoting bizarre beliefs and methods (like articles over the years by these authors which got even more horrifying by the second article in the series).
As I said, I'll be on vacation to catch up on other interests for the next few weeks. Along the way, I'm sure to spend some good "quality" time with beloved albums in front of the sound system as well :-).
Enjoy the summer, enjoy the music everyone...
Great article. What are your thoughts on the popular Computer Audiophile approved build computer vs a regular Windows laptop as a transport via USB to a DAC? I often read on the CA forums how their server build is the best sounding way to do computer audio, but there is never any objective data. Some things those builds do have going for them is small size, quiet, fanless and so on, but I question their claims of superior sound
ReplyDeleteYou have summarized it nicely.
DeleteThey use good components, achieve important goals like low physical noise and I'm sure reliability is good. I see they often use "audiophile approved" USB cards with extra filtering for example which I think isn't a bad thing along with lower electrical noise power supplies.
As much as these things may improve subjective sense of security that one has "the best", I do not feel they likely have significant sonic impact. I've certainly tried many computers over the years (Windows and Macs, towers and laptops) and did not think they change the sound quality of a reasonable DAC nor have I seen measurable differences (see again the review of the AudioEngine D3 last year). I too would like to see some evidence for sonic changes. Until then, I see the testimony around audio sounding "different" or "better" as being unlikely...
Neat! I'm very interested to know about Linux x Windows and if ALSA/PulseAudio output will make any difference. I just listen "different" when I tested Linux, maybe placebo, but my perception "says" the Linux with ALSA have more solid sound than windows.
ReplyDeleteGood point Guga.
DeleteI unfortunately have not had much chance to play with Linux over the last decade as my work is too dependent on Windows / Mac applications. As such I must admit I don't know what PulseAudio is. I do have a little NetBook installed with a recent Ubuntu which I might have a look at! If you have some tips about the best Linux OS to look at and apps for audio playback, let me know and I'll see if I have an opportunity to try...
I think the obvious questions would be whether the drivers will work reliably for my DACs like the TEAC. And whether they will achieve bit-perfect data transfer.
I don't know anything about Linux, but Ubuntu is one of the distros that will work with HQPlayer. This is another audiophile playback software and costs a princely $100. Of course it's another software player touted as sounding great. If you get a chance I wouldn't mind seeing tests on it. There is a 30 day demo trial.
DeleteOn Linux, the name ALSA encompasses the kernel drivers and the low-level userspace libraries. On its own ALSA provides only the basics, and most systems run something like PulseAudio or JACK to complete the audio stack. These systems perform automatic resampling and mixing to allow simple playback from multiple simultaneous sources and in formats not directly supported by the hardware. If you're looking for the highest quality, you'll want to bypass these layers or put them in some kind of passthrough mode such that you get bit-perfect output. Fortunately, this is easy to do. With both in bit-perfect mode, there is no reason Linux would sound any different from Windows, though I'm sure plenty of people will disagree.
DeleteThanks for the info on Linux guys.
DeleteDoes ALSA handle USB Class 2 audio natively like on Mac for the higher sample rates? That could really make life a lot easier I would imagine! Maybe I'll plug my TEAC into the old Ubuntu netbook to have a look when I have time! Another distro I've heard is good is Mint Linux - more traditional desktop (GNOME I think). I might give that one a try also!
HQPayer is interesting in that it does quite complex upsampling to PCM/DSD with various algorithms including the different filters we've tried here in the past as well as DSP functions like FIR room correction and EQ. In that way, it *does* actually do something unique and does not claim "bitperfect" output; the likes of JPLAY.
ALSA should support USB Audio Class 2 though I don't have such a device, so I can't confirm it.
DeleteYes, ALSA does natively support USB Audio Class 2. There are number of Linux distros optimized for audio playback that leverage ALSA. My favorite is Volumio. Have it running on a Raspberry Pi B+ and it's awesome. Currently using an HRT MusicStreamer II+ DAC with it, however about to receive an IQAudIO Pi-DAC+. It's an I2S DAC that connects on top of the Pi. Worth trying for $46, and it has hardware volume control with ALSA.
DeleteA friend of mine, a lifelong audiophile, has a really high end audio setup, including some esoteric DAC that I think is somewhere in the $5000 - $10,000 range. The source of his music is an old MacBook Pro, with a USB cable attached to his DAC. His music library is AIFF, and I believe he uses Audirvana for playback.
ReplyDeleteHe disconnects the power cable to his laptop when playing music, as he's found the sound is better when his MacBook Pro is just running off the battery. His belief is that when the MBP is being charged there is noise leaking through the USB connection to his DAC.
Have you done any measurements of playback from laptops A/B'ing a powered connection versus running on a battery?
Hi Jim.
DeleteIndeed I have heard those claims as well over the years. For times I have tried, I have never heard or measured a difference between A/C and battery using a laptop.
I have tested this a number of times and not found a difference. I think the one time I published some graphs was last year with the AudioEngine D3 DAC measurements because this is a USB DAC powered off the interface so I figured if there is something to be found, it would be these kinds of DACs.
Result here:
http://archimago.blogspot.ca/2014/03/measurements-audioengine-d3-usb-dac.html
Check out the "II. Comparisons" section where I listed "24/96 AudioE D3 (AC)" and "24/96 AudioE D3 (Bat)" measurements connected to an ASUS Taichi Ultrabook laptop. No difference. I didn't publish the results with the MacBook Pro but this was the same finding.
Whether there is noise getting through from the A/C charger or not, I basically have not seen it be a problem by the analogue output stage of the reputable DACs I have tried with measurements. IMO either it's not a big problem or the DACs are already doing a good job filtering it out; at least in the frequency range that we care about in audio!
Thanks for pointing me to that posting of yours Archimago.
DeleteWell, the upside of my friend upgrading to his uber-expensive DAC is I have his HRT MusicStreamer II+ on permanent loan from him. I picked up a Raspberry Pi, played around with a few linux distros, and settled on the Volumio platform. It's an awesome setup, connecting the Raspberry Pi USB output to the Music Streamer and then on to my Bryston pre-amp.
There was an interesting comment on an Audio forum by Kevin Halverson, the designer of the MusicStreamer, responding to someone who claimed it would sound better if battery powered. He claimed that his DAC has a completely isolated power stage, so even though it is "USB powered", powering it via battery would have no effect on sound quality:
http://www.head-fi.org/t/480567/v-dac-vs-hrt-music-streamer-ii#post_6688298
Great that I stumbled upon a setup that is low cost but sounds better than my beloved Squeezebox Touch.
Nice Jim! Love it when friends (and family) are so generous :-).
DeleteInteresting comment from HRT. It certainly suggests that at least with this device all those recently released USB filter devices are not necessary (stuff like the AQ JitterBug and Regen...)... Hard to know I guess unless one tests them out in the system but without the manufacturer publishing data on what they do exactly, it's hard to understand as consumers in what situations, for what anomalies, and to what magnitude they affect devices like DACs.
Raspberry Pi - there's another device I might get my hands on! Very inexpensive, low power, and I'm sure tuns of fun to poke around with :-).
If you do pick up a Raspberry Pi (and why not for $50) there are a bunch of add-on I2S DAC's for the Pi that a lot of people are having fun with. And definitely try out the Volumio distribution (https://volumio.org) - it's a total blast.
DeleteAudio processing is simply trivial on today's low end hardware. You have AQ stating such absurdities as more powerful hardware yields better sound, and for Windows based PC's stay away from Kingston RAM modules.
ReplyDeleteBottom line is a $49 mainboard with AMD Kabini QuadCore won't even come near breaking a sweat.
The C.A.P.S idea at computer audiophile is a joke. I'm all for running quality components but until they take a computer with simply a clean installation of Win8.1 and show the difference in a blind test they are simply howling at the moon.
Thanks for the laugh. Exactly how I thought about it!
DeleteTo the point :-). I like it.
DeleteArchimago, thanks for the article. It's painful reading the "professional" audio blogs. I have a reasonable level of experience with enterprise computing and networking - perusing computer audiophile forums is a hazardous occupation.
ReplyDeleteI truly appreciate you sharing your efforts and musings with the world. I admire your thoroughness and quest for factual understandings. There should be more like you writing professionally.
Thanks Qmax. I do hope the "professionals" will in time change their viewpoint on many of the current "audiophile dogma" which I believe rings extremely hollow to many of us. Their insistence on certain beliefs and lack of apparent ability to express skepticism "kills" the hobby in time and isolates those with more objective, scientific mindsets. While I know many who stay rational, those who persist in pursuing unrealistic beliefs become more "idiosyncratic", eccentric and marginal in the eyes of mainstream music lovers and tech enthusiasts. Is it any wonder that mainstream general tech websites criticize $300 ethernet cables?
DeleteThe problem I believe is the fact that professionals have to make money. How else to get fancy gear in to review on loan? Increase the website hits and newsstand interest? Collect advertising revenue? If one wants the business of a cable company, what else to do but try to drum up business and interest even if the review isn't whole-hearted... At least support claims that it "could" make things sound better, right? (Remembering of course that general tech websites derive advertising from all sorts of manufacturers and not just the "specialized" audiophile ones so they probably don't mind to piss off a company like AudioQuest with a cable dissection as a recent example.)
Thankfully, I am free of the need to deriving income from what I write here.
Just stumbled upon this page, which in great detail describes what I surmised from reading a large number of online audio publications over the past year:
Deletehttp://www.high-endaudio.com/magaz.html
Especially like Secret Rule No. 8- "The more dishonest your magazine is, the more you shall proclaim your honesty." Explains why Michael Lavorgna at Audiostream goes ballistic anytime anyone suggests his writing might be influenced by his advertisers.
I am not surprised your tests failed to show measurable differences.
ReplyDeleteA: You are biased towards NOT (willing) to hear obvious differences and thus choose tests accordingly to prove your point.
B: You are obviously not using audiophile (USB) cables.
C: Your DAC is rated very low by audiophiles and is unable to show the differences.
D: You did not use USB signal improving 'gadgets' (USB regen, Wyrd, or USB filters).
E: The USB port of your PC is equally crappy regardless of OS (both jitter/timing/noise/+5V).
F: Null tests are flawed because the clock is not synchronised.
G: Your transducers are NOT revealing enough.
H: Your amplifier does not contain tubes nor audio transformers, a MUST for high end as semiconductors remove the 'musicality'.
I: None of your components have been modified, they are all stock and thus do not perform optimal.
J: J-test does not show the real jitter, the test is flawed.
K: It's computer generated noise in the USB/other outputs that 'somehow' makes it in the analog path but NOT the loopback path.
L: You used the wrong recordings/source/ripping program/transport and pay the price by not being able to hear the differences.
M: RMAA is a worthless and non-revealing 'tool', it should be banned.
N: You did not use audiophile power conditioning and your airco or CFL lighting ruined the tests.
O: You should have invited Lavorgna or Fremer for the subjective tests (or equally qualified guru)
P: No serious listener would ever include MegaPoop3 as test signal.
Q: The EMU 0404 ADC is outdated and cannot record that what can easily be heard, its drivers are poor.
R: You have used the wrong brand of SSD storage (obviously, cause mine sounds very different).
S: JPlay obviously sounds much better than other players, measurement signals, however, cannot show this as they are too 'basic'.
T: Bits are NOT bits, no matter how you think about it. It's the biggest lie in audio these days and not recognised by the industry but by a few people.
U: Timing is everything, your system simply isn't good with timing, otherwise it would become ever so clear.
V: You did not use 'balanced' signal paths ... just single ended, SE is unable to resolve the obvious differences.
W: You have too much time on your hand and should only use that to listen, not play with pointless measurements.
X: The set of measurements you use is unable to show what the ear can easily hear.
Y: Do you do these tests ? (HINT: You must include the alfabet letter Y for it to be funny)
Z: ZZZZZZ... your tests and conclusions are boring and pointless, just fell asleep.
Sorry, ran out of alfabet letters .. it's possible I left out a number of other reasons though.
Have a good vacation and keep doing what you do.
Some counterbalance to subjective 'power talk' is needed as well..
LOL! Jeez man, that's some awesome creative writing :-). You should write a "compendium of audiophile rationalizations" I think!
DeleteI was going to say that Archimago would loose the wrath of Lavorgna, but you beat me to it!
DeleteSolderdude, your discourse, aside from a few spelling and grammar issues, appear to indicate a fair depth of knowledge on both the equipment and methodology needed to "properly" conduct such testing. May I suggest you perform such testing to your parameters and post the evidence and results to show how wrong the test(s) at hand are.
DeleteEnglish is not my native language which explains the spelling and grammar issues... I hope.
DeleteYour assumed depth of my knowledge is probably not warranted.
I was just having a bit of fun at the expense of certain types of audiophiles ...
I hope you weren't offended by my tomfoolery.
Wow... nice rant.
ReplyDeleteKeep up the good work Archimaga, it's clearly having an effect!
Have you tried including Win 7? Had a day experimenting. Simple A/Bing not as grand and scientific as yours, just same setup, same software, different OS.
ReplyDeleteLike you I found Win 8.1 and Win 10 sounds the same (to my ears). However I liked the sound of Win 7 Ult. x64 more. For me Win 8.1 and Win 10 is a bit "liquid/wet"??
Of course there is the driver thing.
Hoping if you include Win 7 on your tests it would be more conclusive...
Thank you!
Hi Michael. No, it has been a long time since I had a machine running Windows 7.
DeleteI think it is important to try generalizing these findings and say with *good confidence* that OSes in general DO NOT make a significant difference for modern asynchronous USB DACs. As outlined above, so long as bit-perfection is achieved, the timing is handled internally by the DAC, thus off-loading this variable from the main computer itself. This is also the reason why bit-perfect player software and these OS optimizations make no difference. Even if you change scheduling, priorities, assign "realtime" levels of response, it will not translate into anything that the DAC does so long as the buffer does not end up empty of audio data...
If one is arguing for noise level to decrease from software like JPLAY and the OS tweaks, then I don't have any evidence there either that anything has changed since the noise levels I'm seeing out of the DAC are already better than a 16-bit CD noise floor with stock Win 10. In fact, we could argue that Fidelizer and JPLAY could be making noise worse by the insistence to use smaller buffers which make the computer work harder! (I've honestly never understood why the insistence on this.)
As in the other links to tests previously done (eg. AudioEngine D3 measurements), I have demonstrated this "truth" among different computers and even very different OSes like Windows vs. Mac OSX. Therefore it is unlikely Windows 7 would present any special or particular situation since it basically is an older version of Win 8 / 10, even down to the same driver architecture...
Yeah, crazy are these things. Its just like USB cables. Don't know what to believe. But the bottom line were here to enjoy the music. And its and still summer! Enjoy!
DeleteMichael
Fidelizer optimizes OS and MMCSS stuff so software buffer is irreverent to Fidelizer. You shouldn't mix Fidelizer and JPLAY together to cause misunderstandings.
DeleteWell, so much for all these scam tweaking softwares. It's the digital placebo for the enthusiast computer audiophiles. You tweak here and there, put and overclocked Hashwell-e I7 with 16GB of RAM and some NAS with SSD for "smoother sound" and there you go, the BEST SOUNDING PC EVER! I'm just waiting fot the day we will see some "audiophile" HD's, SATA cables and etc. Make and ordinary HD, stick one "HI-RES audiophile military class" logo on it and increase the price in 1000%!
ReplyDeleteThis is too much for me, haha!
BTW, you wrote that your Pentium is running completely fanless. You don't have ANY fan in the entire HTPC? Oh boy, i'm aiming for that, but as a gamer too, i need some quiet fans actually. I'm looking at the new Intel Skylake, especially the not-yet-released I5/I7 versions with 65w and 35w of TDP. These combined with a "tower" CPU cooler and a "0 DB" VGA (passive cooling or some cards that only spin its fans in high temperatures while you are gaming) will result in a well balanced HTPC, completely silent for music, but powerful for some action ;)
As always, really, really, really nice article!!
Enjoy your summer and your vacation!
Best regards!!
PS: since we are talking about computer audio, make some tests with HDMI audio from the computer (motherboard or AMD/Nvidia VGA) when possible. HDMI is like a villian among audiophiles, but i think that it's not that bad. Maybe it's as good as many budget USB DAC's. Or maybe better than that in some cases.
Hi VK!
DeleteThe problem with audio is that it is so "ephemeral"... The most unreliable in terms of memory retention thus the most difficult to really get a handle on whether something is "better" than another. The best we can do is simultaneously play 2 sample side-by-side and flip instantaneously hoping we can detect these subtle differences... Hence the need for multiple comparisons through ABX trials with the hope of statistical proof that we can/cannot discern a difference. Unless of course the sonic difference is easily audible (not usually the case with high-end audio since I hope the baseline is excellent quality already!).
Already you can find these JCAT SATA cables (a *steal* at >300Euros) and stuff by the purveyors of JPLAY ;-).
Yes, my Pentium is is fanless. Haswell Pentium G3220 running at 2.8GHz (3GHz stock), around 0.95V or so undervolted bolted to the big heatsink. Fanless power supply. It's in my basement here in Vancouver so it never really gets too hot ambient temperature down there. No fans at all running in the computer. No way I'd be playing games on that though. Yup. I'm looking forward to Sky Lake also. The future for me is these processors fanless so long as the graphics is capable of HDMI2 for 4K/60Hz - I want a silent 4K capable HTPC!
I have already measured my ONKYO TX-NR1009 receiver through HDMI and its SPDIF a few years back using the pre-amp outs! Overall it wasn't as bad as I thought... The DAC component was capable of ~18-bits resolution (!) and the HDMI had some jitter. It was intermediate between TosLink and coaxial. So bottom line was that I found the receiver more jittery than a dedicated audio DAC and HDMI seemed about the same jitteriness as SPDIF in this device. Also, I could measure a small drop in resolution when decoding 5.1 vs. pure stereo. That was done in 2013 and this receiver came out in 2011 I think so perhaps things have improved? Note that the test was with my AMD A10 "APU" machine thru HDMI at that time and not the Pentium G3220 I used in this post...
Test:
http://archimago.blogspot.ca/2013/10/measurements-onkyo-tx-nr1009-as-hdmi.html
Also HDMI cable tests:
http://archimago.blogspot.ca/2013/11/guest-review-measurements-quantum-hdmi.html
I wonder if the HDMI license is too expensive for small audiophile companies. A shame because I am a fan of multichannel and it'd be great to have more inexpensive high quality multichannel gear like DACs. I did wonder about the use of DisplayPort instead which is a VESA standard though unfortunately not as common in home AV.
Thanks for the response!
DeleteGood to know that HDMI has similar audio quality than SPDIF. Thanks for the feedback. It's hard to find unbiased measurements of HDMI audio vs anything else (it's always some kind of "HDMI is a junk and i will prove it in this test of this HDMI AV receiver vs $ 500k top DAC"... why so much hate of HDMI??).
And yes, the main problem with HDMI is the royalties. And about the Display Port i agree. It'll be a nice addition to AV systems. Unfortunately we have so much good ways to connect our gear, but the lack of a single standard and constant "standard upgades"complicates everything (HDMI 1.0 - 1.2 - 1.3 - 1.4a - 2.0 and what to say about USB with it's constant changes from the popular and well accepted 2.0 to 3.0 and now 3.1 type A and C AND with Intel trying to bury everything with Thunderbolt... oh boy, i just want to keep it simple with LESS cables, and not more...).
All the best for you!
hopefully your readers realise what a result down at -112dB means , your "lowfi" ;) stuff resolves beyound CD resolution and nothing shows up there . The implication, anyone "hearing" a diffrence with 16/44.1 files or recordings who's practical resolutions is less than this -112dB, should consider alternative explanations (they are really not alternative lets call them the more probable explanations ) . I mean your own diy random noise vs random noise inherit in a 24/192 file with 70´s rock does it matter. If one wonders why lossles compression is less effective on some hirez files one should consider that compression relies on patterns and predicability random don't compress very well.
ReplyDeleteGood points. It is impressive what typical audio equipment is capable of these days. And it's sometimes easy to forget the magnitude of the dB scale as a logarithmic entity!
DeleteThere are many remasterings of old recordings coming out in hi-res these days like those 70's rock albums, 50's and 60's jazz. Certainly hard to imagine 24-bits or 96+kHz making a big difference in themselves... But of course the quality of the mastering itself is likely the reason for improvements heard.
I suppose at least these old remasters have good dynamic range... As opposed to DR6 modern recordings - seriously, Bon Jovi "Burning Bridges" in 24-bits!?
Oh yes I heard both bad and very good remastered recordings regardless of delivery format. My point was that even the recorded noise will swamp imperfections in your modest setup :) and I did not even mention ambient noise in the listening environment . So if by some very remote chance some of the stuff tested does something to the music beyond your measuring capability no one will hear it anyway . If the problem is below -112,5 dB it is by all normal practical means completely inaudible . Even with the best hifi ever and in an completely silent room the recorded noise will then further cover it up . I have tested the resolution of my own hifi ( Meridian ) it is hovering around 112dB at the speakers ? What did I have to do to observe that I have a test signal at -112dB with a test tone in the upper midrange , I can barely hear it trough the amplifier noise . IF I turn the volume to max AND press my good ear against the drivers in the speakers , with that volume normal recording noise had been audible across my whole apartment . A humbling test to do is to have test signals like -60dB -70dB etc down to -100 etc and keep your volume at your normal listening level in your normal listening environment ( with ambient noise and all ) and then actually hear at what surprisingly high levels you can't hear anything at all anymore . Then you crank it up to max and you will still get surprised and then finally you try the ear to the driver method :)
ReplyDeleteIndeed the -70dB 'floor' is also what I have determined for myself long ago.
DeleteAs the story around this subject is too big in size to post here I had written some things about DR, noise and hearing here:
https://diyaudioheaven.wordpress.com/tutorials/dynamic-range/
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Irrespective of the softwares like Fidilizer, Audiophile Optimizer etc., some basic tweaks are a must to get good audio from PC irrespective of whether using a high end asynchronous outboard DAC or the on board PC soundcard.....
ReplyDeleteOne needs to bypass the windows mixer and use WASAPI or ASIO, this makes a huge and evident difference in sound quality and also employs any software based tweaks as advised by the developer.
A noisy PC will add to the problems, a fanless design with a linear power supply also helps a lot in keeping the noise level well under the decibel levels.
I think rest of the tweaks add if at all only an infinitesimal (or should I say inaudible) level of improvement.
But an audiophile's journey never ends.....so they say....
Thanks for the valuable insights. But please, could you repeat your tests using the Linux based daphile (www.daphile.com) audio os. You have an option to launch daphile from a USB stick after which it runs in memory. It won't overwrite the system disk so your original windows setup is preserved.
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