Saturday, 23 October 2021

Miscellany: FlexASIO for 384kHz, Philips 2001 SACD/DSD64 test signals (thanks Black Elk!), and Roon network multicast. [And Coltrane's A Love Supreme Live Hi-Res, Music Industry Crystal Ball, Magic Quantum Fuses!]

A scene from The Simpsons 1996...

Hey everyone, for this week, let's talk about a few "miscellaneous" topics which I've either wanted to mention over the last few months or have just come up as interesting tidbits I think worth documenting but not necessarily large enough as topics in individual posts. The main topics are:

A. Instead of ASIO4All, we can use FlexASIO with the E1DA Cosmos ADC for 384kHz samplerate measurements in Room EQ Wizard.

B. A look at some "standard" SACD test signals from Philips back in 2001. With many thanks and great discussions with Black Elk.

C. Roon needs network multicasting. Check this out if you're running into network issues; I had some problems initially with my ASUS ROG GT-AX11000 and relatively complex home network.

A. FlexASIO, up to 384kHz using REW and ADCs.

Instead of ASIO4All, I saw a tip on E1DA's site to try out FlexASIO which will allow operations up to 384kHz (instead of 192kHz max). Yup, the current 1.8 version works quite well and go install FlexASIO_GUI v0.3 while you're at it.

A quick peek at the control panel for set-up:

FlexASIO is a "wrapper" that will allow you to use the ASIO interface with devices through one of the other driver types on your computer system. In Windows, there's the option of using WASAPI, WDM/Kernel Streaming, MME, and DirectSound (ASIO4All uses WDM/Kernel Streaming). I've come across situations where the driver refuses to load due to parameters being incompatible; for example, I've had issues using the WDM/Kernel Streaming setting with the E1DAC Cosmos ADC. In these situations, you can manually delete the configuration file:
C:\Users\<User name>\FlexASIO.toml

And restart the driver config in REW. Or use the FlexASIO_GUI to change the settings.

With those settings above, here's a look at a 1kHz -3dBFS signal from the RME ADI-2 Pro FS XLR, +19dBu level (the older AK4490 DAC model, not the more recent Pro FS R Black Edition) to E1DA Cosmos ADC. With this DAC, -3dBFS yields about the highest THD+N result:

There you are, frequencies all the way to 192kHz. We can of course look at basically any signal of interest including for example the "1/10 Decade Multitone 32":

Within 15Hz to 30kHz, we see better than 118dB distortion/noise-free range (>19-bits) on the ADI-2 Pro FS.

While this works, of course I still think it would be nice for REW to support WASAPI natively, and prefer that the E1DA Cosmos ADC's ComTrue ASIO driver worked in REW!

B. A look at Philips' 2001 SACD DAC Test Disc ("Ichimura & Sekii") - Scarlet Book 1.2 standard.

I mentioned in the previous post using SoX-DSD that I would start using that to encode my DSD test signals, primarily "sdm-8" noise shaping as my "standard". Reader Black Elk who in the past had worked with Sony at the dawn of SACD was able to provide a few ripped samples from this disc for me to examine:

Cool! An actual test disc from one of the creators of SACD based on "version 1.2" dated May 2001 of the standard (you might recall the term "Scarlet Book" referring to the SACD specs). When ripping the DSD64 data, the CD-TEXT disc title was labelled as "Ichimura & Sekii" which I assume must be the names of the engineers who created these synthetic signals; you'll find both these names either in the DSD literature or Sony's 1-bit digital patent.

As you can see in the disc contents, there are various test tones on the disc. So let's compare this "SACD standard" with SoX-DSD sdm-8. First, let's have a look at the noise floor:

Based on the filename, the Philips Test Disc track appears to be using a 7th-order filter. The noise level is similar through the audible spectrum, with the sdm-8 setting keeping noise lower a bit further out from the audible range until 25kHz. The Philips encoding allows noise to rise relatively quickly just above 20kHz. However, notice that the Philips noise floor appears just slightly lower in the audible range compared to SoX-DSD sdm-8.

Let's compare the 1kHz 0dBFS tone (I encoded the SoX-DSD at -0.1dBFS to prevent any clipping):


Interesting. As per the noise floor comparison above, we see that indeed the SoX-DSD encoded signal pushes ultrasonic noise a bit further out. It has better THD, but lower THD+N on account of the "N" component with a slightly higher noise level than the Philips v1.2 Test Disc signal.

All in all, we see differences but we're looking at just +/-1dB variations with better than -100dB THD+N so the variance would not really be of significance for music. Note that this is with an actual DAC playback. We could just analyze the signal in the digital domain with DSD-to-PCM conversion in the computer; but that's not as fun as playing it on an actual DAC. ;-)

Here's the Philips SACD Test Disc encoding of the 19 & 20kHz tones, commonly used for intermodulation (CCIF/ITU-R) testing:

Looks clean through the ADI-2 Pro FS. We're looking at about -116dB on that 1kHz "d2L" intermodulation product.

It's quite possible that similar signals were used by John Atkinson back in 2001 when he reviewed the Sony SCD-1 (he talked about using a "provisional Test SACD" from Sony) and got measurements like Figure 15 which is a 1/3-octave smoothed graph using the -60dBFS and -120dBFS 1kHz signals. I see these on the Philips Test Disc as well.

I don't know why Atkinson chose the low-resolution 1/3-octave smoothing back then with the Audio Precision System One; maybe limitations of the resolution and rising noise floor with that equipment when looking at bandwidth >20kHz.

These days, more than 20 years later, we can just grab a hi-res 1M-point FFT and show the accuracy of the DSD signal, 192kHz frequency bandwidth, no smoothing:

No harmonic distortion of concern as expected at these low output levels of -60/-120dBFS.

Thanks again to reader Black Elk for sharing these Philips Test Disc signals from the early days of "hi-res audio". It's great virtually "meeting" folks like him with backgrounds and experiences in the technology over the years through this blog.

We also talked about this 2002 "SACD Surround Sound Reference Disc" which he worked on:

With his permission, this is what he said about it when I mentioned that the disc had some great-sounding tracks on it:
"... It was largely put together by ex-Philips Classics balance engineer/producer Hein Dekker. His own recordings are excellent, but Philips Electronics developed close relationships with a number of top Dutch recording engineers (Ronald Prent for rock/pop; the team at Polyhymnia [formerly Philips Classics Recording Centre] but especially Erdo Groot). They recorded most of the stuff on the disc, and it is reference/demo quality.

One of the many missed opportunities with SACD came with the recording of works by Jerry Goldsmith. While the recordings on that disc (which were released by Philips on some promos, and later by Telarc) are good, they could have been so much better. They were made at Abbey Road with legendary engineer/producer Bruce Botnick (The Doors and a gazillion others). I love Bruce, he's a really sweet guy, but he and Jerry were used to working on major movie soundtracks (either in LA or London), and they opted to work at Abbey Road with Bruce's tube mics.

However, those recordings only came about by chance. We were in Budapest with Dekker (2-ch) and Groot (M-ch) to record Dvorak symphonies with Ivan Fischer and the Budapest Festival Orchestra. We knew the orchestra was going to be rehearsing with another conductor, so the guys were only able to set up their main pick-up (for 2-ch that was 4 mics across the front of the stage, and for M-ch a 5-mic array at 0 deg, +/-30 deg and +/- 110 deg. The two mics. at +/-30 were common to both microphone arrays.

As the orchestra layout was going to be completely different, they could not rig any spot microphones. With the limited work done, we decamped to the M-ch control room, which was the old control room set up by Hungaroton. We could hear the orchestra tuning up through the open door, and Erdo said, "Well, we may as well listen to what they are going to rehearse." So, he opened up the 5 microphones on his array.

As soon as the orchestra had settled they launched into the theme from Star Trek (the movie scored by Goldsmith, but containing part of the TV theme). That got our attention, and, after the piece broke down, we went into record and recorded the rest of the rehearsal. It sounded amazing. Just 5 mics., and the orchestra in a concert hall with a natural acoustic.

When they had a break Dekker ran down and introduced himself, explained what had happened, what we were doing with SACD, and invited Goldsmith to come hear a playback, with the condition that we would erase everything on his say so.

We played a few pieces with him in Erdo's chair, and I remember watching his face and thinking, 'Uh-oh, he hates it.' but he loved it. In subsequent breaks Dekker thrashed out a plan to record all the pieces we heard, but he opted to do the work at Abbey Road with the LSO instead. Abbey Road doesn't have the acoustics of a concert hall, and Polyhymnia have customized all of their microphone electronics to make them state-of-the-art, so we lost a lot when we did it for real.

I think Gus Skinas still has some of the raw recordings in his archive. They are fantastic 'you are there' recordings, but also you hear the musicians pick up and drop things and make noises since they were not in a recording situation.

Still, we had some fun demo'ing Basic Instinct at shows! :D"
As for DSD beyond DSD64:
"With regard to high speed DSD, I don't know if anyone is recording at anything higher than DSD256, and there's likely only a handful of people recording at DSD128 and DSD256. I'm sure DSD512 and DSD1024, etc. are just gimmicks like the oversampling factor on CD players back in the 80s. DSD designers like Andreas Koch (Playback Designs) and Ted Smith (PS Audio) will tell you that you get nothing for nothing, and while DSDX may look great on paper in practice you may hit other problems due to the higher clock jitter requirements. Andreas thinks DSD128 is the sweet-spot, while Ted sees some benefit to DSD256 (can't remember exactly what now). Just like in the days of PCM DDCs (where users would upsample all CDs to 24/192) there may well be audible differences, but you may not be getting exactly what you think you are getting."
Yup, that certainly resonates with the technical data. I think DSD64 if we're looking for best fidelity is a bit too "tight" in terms of the rising noise level being close to the 20kHz upper potentially audible range. DSD128 is better, and DSD256 would be IMO the highest that we should need assuming one has a good DSD playback DAC.

As for high ultrasonic noise in DSD64:
"Sony fried a few high-end amps in the very early days of DSD because they had no output filter (Philips must have learned their lesson when they introduced Bitstream DAC based players in the mid-80s). Tube amps also didn't usually respond well to the SDM noise. We used to term this distortion 'birdies'."
So the "technical" term is "birdies", eh? ;-) As tempting as it is to leave the signal alone and not apply any filtering, I agree that with a DAC like the RME ADI-2, I'd keep the filter at 50kHz rather than 150kHz for listening just in case that DSD noise doesn't play well with the amp! 

Great to hear these kinds of background insights from folks who were "there"! And the discussion was too good not to share with you (with BE's permission of course). ;-)

As audiophiles, I think more and more we need to be educated with the experience and know-how of the professional audio world. We cannot remain isolated in a little "bubble" of audiophilia talking about often questionable beliefs ("myths" even) when in truth what we are listening to are products of the audio engineers out there working with state-of-the-art equipment and the actual artists themselves!

If you want to read more from Black Elk, make sure to check out this Steve Hoffman Forum thread on early SACDs with great insights and anecdotes. I didn't know about the "visual watermarking" feature that was promised for example, and this is the best summary of what happened to SACD within Sony/Philips I have ever read.

C. Roon needs multicast.

Here's an excellent post on the Roon forum discussing lots of details which you might find very interesting. This issue with multicast came up when I was setting up my ASUS ROG Rapture GT-AX11000 back in April.

I was noticing that on occasion, some of the Roon endpoints on my network would randomly stop playing. Depending on the time of the day and how many devices were on, the stoppage could be after an hour or within minutes of playback. Here's the network architecture diagram at my home as a reminder:

I run RoonServer on my Windows 2019 Server machine and I can check the Roon log at:


Every time the playback stopped on whichever endpoint, it looked like an error of this nature:

Notice the "Unable to read data from the transport connection" issue. Well, the way to resolve this with the ASUS router is to turn on multicast which on mine defaulted to being off:

If you have a managed switch, you might also want to check out the multicast settings. Turning on IGMP snooping and blocking unknown multicast address to reduce denial-of-service (DoS) attacks might be something you'll want to do as well. So for example, here's my "audiophile network switch", the Netgear Nighthawk S8000 settings:

Before we throw caution to the wind and turn on all kinds of network parameters, note that there are some which could shutdown the communication between Roon and endpoints. For example on my mixed 10/2.5GbE QNAP QSW-M2108-2C switch, don't turn on "Multicast flood blocking" or this would prevent the Google Chromecast and AirPlay devices (like Yamaha receiver or AppleTV) from being seen by the Server:

It's good to be systematic when dealing with network issues, changing one parameter at a time so as to discover nuances like this.

One other thing to check with Roon is to make sure the Windows Firewall is allowing the RoonServer to get through:

As you can see, "roon.exe", "RoonAppliance", "roonappliance.exe", "RoonServer" have all been allowed through. Make sure you also check for "RAATServer" and "raatserver.exe" items on the list.

One last thing I checked on my network was latency since I allow large buffers for the 10GbE connection, especially between my main Workstation computer and Server. You can do this with something like "ping -t <RoonServer computer address>" (my server is at which will continuously ping the computer every second and you can get a reading of latency while doing stuff like playing music streamed from the Roon server, or doing something like a large file transfer as shown below:

In the picture above, I'm doing a massive 70GB file transfer between the Roon Server computer to my Workstation while I'm also playing music through the Roon app. Notice the file transfer between SSDs on the two machines is running at 340MB/s (the limit is the SSD read/write speeds, not the 10GbE network which is even faster). The "ping" looks good, <1ms with the highest result being merely a single 2ms round-trip latency throughout the whole 3+ minutes large file transfer.

High "network jitter" will show up as variations in that "ping" time when the system is busy and not responding quickly. If you see significant issues, you might want to play with the transmit and receive buffers on your network device driver.

For the record, with the Windows 2019 Server computer (32GB RAM), I've set it at 2048 receive and 4096 transmit buffers these days on the 10GbE ASUS XG-C100C card - a 2:1 ratio for  transmit:receive buffers has been said to be optimal for many machines; well, it works for me even though I haven't done any systematic testing. Be mindful that an excessive number of buffers could worsen latency in the form of "bufferbloat". Smaller buffer sizes like 512 and 1024 would be fine with 1GbE networks.

Settings above have resulted in very stable playback with Roon through my network these days across various audio endpoints. No problem with concurrent streaming to 3 or 4 devices around the house. No more "Unable to read data..." log issues.

Check out Roon's "Networking Best Practices" page for more if you have issues.


My new album this week is John Coltrane's A Love Supreme: Live In Seattle (1965) (2021 Release, DR13). It's available on most major streaming sites (no Tidal). An interesting historical live performance (see Rolling Stones review) and worth a listen if you like A Love Supreme (studio recording). Note that this album is definitely not for everyone, nor is the "avant-garde" jazz genre in general appreciated by many. ;-)

For those thinking about the hi-res 24/192 version, I would not bother:

As you can see above (Track 1), as an old analogue capture, this is not a high-resolution recording with relatively high noise floor and ultrasonic stuff of high levels. Notice the -35dBFS 77kHz tone for example which sometimes will go up to -30dBFS like on Track 3. IMO, as discussed before, it's best to discard this stuff and potentially improve sound quality plus save storage space. Just get the CD or downsample to 16/48.

Coltrane died about 2 years after this recording in 1967 at the young age of 40 from liver cancer.

A friend sent me an interesting blog piece the other day titled "12 Predictions for the Future of Music" by Ted Gioia who has written a number of books about music history (especially jazz). Pretty dire stuff which considers music (especially pop music) as a generic, basically disposable, commodity these days driven by price-conscious consumers fed by the tech giants with a tremendous volume of music in existence already (and tens of thousands of albums released each week!). I think the predictions are certainly not unreasonable given the trends we have been seeing not just in the mechanism of music distribution (big tech), but also in the quality of the music these days and the short-lived artiste du jour.

Lots of pretty safe predictions there I think even if worrisome for the music industry and for artists. Items 11 and 12 seem about right as well:
11. Record labels will increasingly pursue a bare-bones, low investment approach. A few execs will be able to maintain lavish lifestyles even in a declining industry, but there won’t be much cash for those at lower levels in the organization. Most people working in the music business will be poorly paid, with few opportunities for advancement. New openings at the top will typically go to family members and close friends of the old school bosses—making nepotism the one record business tradition that will survive no matter what.

12. But the greatest dream of the music execs will be. . . to get out (of) the music business. They will try to sell NFTs or promote audiobooks or finance biopics or sell music-themed apparel or open music-themed casinos, etc., etc.. And who can blame them? After collapsing the economics of the record business, they clearly need a new field to destroy.
As dire as the direction of the Music Industry might be heading, as consumers, we have tons of material to enjoy. More stuff, at lower cost, and even lossless quality than at any time in history.


Psst... Wanna buy some "Magic" Quantum Fuses? Check out the review from the shaman, and follow-up from the supposed skeptic(s). Wow, is this some kind of "good cop / bad cop" tactic to get fence-sitting audiophiles to fork over >US$700 on some purple magic fuses?

Need I remind everyone that fuses are passive components (like cables/wires) and it's perplexing how a few centimeters is supposed to help in any way improve sound since it's not even located along the audio signal path and typically in the (directional?!) A/C power circuitry. Let's see the "(quantum) science".

Remember the old saying: "Do not be so open-minded that your brains fall out". Alas audiophiles, I think we're witnessing a few billion neurons on the floor. Clearly this is a good example of "Class A" snake-oil. I suppose snake-oil salesmen have to eat also!

I don't know who these Stereo Times people are and have never read their reviews, but dang, that's some good psychedelics they've been passing around in the form of apparently directional fuses thru USPS over the last year through the pandemic! I guess that's where Synergistic "Research" is advertising these days.

Hope you're all staying safe and having fun with your favourite music.


  1. -- "I don't know who these Stereo Times people are and have never read their reviews, but dang, that's some good psychedelics" --


    A long time ago I wrote some reviews for that website. I came on board with the proviso I would only write some speaker reviews. I couldn't in good conscience write about cables and magic beans...

    It was a fun gig.

    And fwiw: there was no pressure to skew a review either positive or negative. So many people assume in the subjective review community it's all some sort of glad-handing and reviewers-in-cahoots with advertisers. While there certainly is some of that no doubt, it wasn't my experience nor that of the many reviewers I knew from some other publications. There is of course still lots of nonsense and dubious inferences in subjective reviews, but I think much of it comes by honestly :-)

    1. Thanks for sharing that Vaal,
      No, I agree, I don't think it's always about Industry coercion or conscious need to speak well of products (although that likely is also present to some extent). I think it's just a matter of psychology.

      I believe audiophiles are gearheads and passionately love the hobby and the products that promise even better performance. So when some guys talk about magic cables and quantum fuses, I think some of them "honestly" believe what they write. No different than someone of a certain "faith" coming to my door just as honestly wanting to "convert" me to believe in such and such I guess...

      But the point remains that many such beliefs require that "faith" be placed in information transmitted to the reviewer rather than an independent, ideally empirically-based testing and understanding about that information. No doubt, when "favours" are given and relationships are forged (for example, it looks like Clement Perry knows QSA's Steven Tsang), then as "nice guys", we would inherently show gratitude for those fuses sent for review regardless of whether they worked or not (or have any kind of "quantum" effect, etc...). And without presumably the ability to perform objective independent analysis, then there's little to go on but one's own sighted evaluation and the power of psychological confirmation bias empowered by the relationship and those remarkable $$$ figures.

      I trust I'm not too harsh or mean-spirited on write-ups like these but one does have to be clear about the inadequacy of these "subjective only" reviews that make claims which transcend rational, even reasonable beliefs when applied to man-made electronics/engineered products!

      As audiophiles, I think to not speak truth against nonsense is also being complicit with at times basically scams which in a regulated industry would have significantly better consumer protection. If audiophiles cannot speak against these things with knowledge and insight, what good are we but "yes men" perpetuating many of the questionable schemes out there?

      As usual, I hope we as audiophiles can be better at being knowledgeable especially when differentiating truth from fantasy. That's how we can make this tech-based hobby at least somewhat respectable. ;-)

    2. One more thing, since I've gone this far with the monologue ;-)...

      Some people might ask: "But Arch, you've never listened to those Quantum Sciences Audio $700 fuses - so you need to try before you can comment/criticize them!"

      Nonsense. And the answer to that is simple.

      The point is that the onus of proof of whether these fuses work is on those who make such claims given how unlikely the claims are. QSA's website is devoid of any demonstration of effect:

      Likewise, Clement Perry, if he wants to be believed, needs to show us with any simple measurement or other testing that in fact "The ease, dynamic prowess, bass articulation, and speed were also outstandingly more pronounced. The system appeared to have more of a jump factor but simultaneously remained eerily quiet and relaxed."

      Surely if all these things changed to make an "outstanding" difference, the fuse would have an impact on the amp's noise floor, ability to handle high levels with lower distortion, even something like improvements in the TIM test for transients maybe? If none of these things change, then at least he could say he tried and we can perhaps better contextualize what kind of magnitude in change there actually is (ie. seems subtle!).

      No guys, we don't "have" to try anything that's expensive and unlikely to be of benefit to express healthy skepticism.

      If a most trusted friend calls us up one day and tells us to buy a new medication that's a veritable "Fountain of Youth" for a mere US$10,000 because he took it and his arthritis is now gone, I suspect most of us would look for evidence first before handing out the credit card information right? Why not do the same with "quantum" fuses even if it's a "mere" $700? ;-)

  2. Interesting digression on SACD history! But to be honest, I still fail to see a strong technical motivation for the introduction of such a format.

    It seems to me that SDM has made their way into all PCM A/D and D/A conversion devices and we can basically have most of its benefits without the drawbacks of the bitstream. Today it is even complicate to determine how a DAC is treating DSD under the hood and wheter or not a "native" DSD conversion is performed (I recall only AKM provided a "DSD direct" mode?).

    If really something beyond CD quality is needed, 24/96 seems the most logical choice. On the other hand DSD has that kind of purist appeal that sells well to audiophiles, given its immutability.

    But I'd like to hear other opinions on the matter! :)

    1. I agree that the SACD history was very interesting - thanks "Black Elk"!

      To me, clearly the biggest advantage of SACD is the capability of surround sound, but I don't really care about DSD as such. I listen mostly to classical music (where SACD is still very much alive), and I have a surround system, so it's a nice fit for me. Also, in general SACDs tend to be well recorded.

    2. Agree Interference and Freddie,
      It is only in the AKM chips that I'm seeing the "DSD direct" approach of bitstream conversion.

      Yup, I too have no "need" beyond hi-res PCM playback support mainly for the purpose of allowing DSP processing with 24-bit and 32-bit precision.

      DSD classical productions typically have been excellent. I think this is just a result of the awareness that if one is committed to recording in DSD and desires to have the best sounding product, one takes time to do it "right" from the start, recognizing that DSP manipulation of the sound requires PCM conversion. Sort of like "direct to disk" recordings.

      Yes, IMO, lossless multichannel SACD (and DVD-A) was the real step forward back around Y2K when these "hi-res" formats were introduced.

      On an tangent, some DAC companies will talk about their products being capable of "DSD native" and similar terms. DAC guys like Jeff Zhu of Kitsune HiFi (Holo Audio) talked about the real meaning a few years back:

      As per that comment, it seems like the term "native" is most instances is referring not to direct playback of the bitstream but that the computer is sending the DSD data to the DAC without going through DoP conversion. Internally, the DAC converts to PCM sort of like the way many Blu-Ray players often convert to 88.2kHz to feed the DAC.

      The only exception I'm aware of is the Holo Audio Spring 3 DAC:

      which has a separate "Dual Resistor Ladder" for DSD playback. Don't know if anyone has measured the output from that or compared the PCM performance from the same device... Given the cost of going through all the hassle, would be interesting I think if a discrete implementation like this significantly beats an AKM chip "DSD Direct" playback!

    3. Just had a quick peek at the Holo Spring 3 prototype measurements:

      Pretty good for a discrete multibit DAC. Not seeing comparisons suggesting any kind of superlative DSD performance though... Then again, there are no miracles here I think ;-).

  3. Hi Archimago! I just found out that my hearing is no longer as keen as it used to be a few years ago, and would like to enlist the help of measurements to detect distortion and noise differences between comparable audio files/tracks. What will I need, and how should I go about it? I have a Motu M2 if that will be helpful