AuraEngine: Perfectionist off-line mega-tap digital filtering - with hybrid-phase transient detection & blending.

As you know, while digital oversampling filters have been around for ages (4x oversampling available back in the '80s), audiophiles probably got more interested in digital filters thanks to Bob Stuart, and Meridian by 2009 talking about "apodizing" filters. DACs started allowing different filter options. And spin-offs like MQA took things too far utilizing poor designs through the 2010's (and still looking for a reason to exist).

Years ago now, I spent quite a bit of time talking about digital filters across articles like this and contemplating things like pre-ringing here. There were also discussions of using the intermediate phase setting as a compromise. And we even ran a blind test to compare strong pre-ringing vs. no-pre-ringing minimum phase settings.

While I think I've exhausted my time exploring such things at the base level, ever-more-precise filtering options have been available for audiophiles to use. HQPlayer has many realtime software-based playback settings including DSD transcoding. Companies like Chord/Rob Watts have taken their stand on using ultra-tap-length linear phase filters (to the point of extreme IMO).

Off-line upsampling of high precision can also be achieved using software like Pan Galactic Gargle Blaster (PGGB) in 2021 -  I see they have a PGGB+ version now with DSD and up to "ludicrous" 256-bit processing.

Well friends, it's 2026 and the other day I was contacted by Toxa Dev about his new ultra-accurate upsampling system called AuraEngine (current version 1.2.2) available on Github; full source code published and freely released for non-commercial use. You can also download and run the pre-built Windows portable app here. Notice that there are three downloads available: Starter 1M-taps, Standard 10M-taps, and Reference 30M-taps. Pick your level of obsession err... weapon against weak upsampling!

Using low-power Beelink Mini S as Linux Roon streamer - HQPlayer (PCM 8x, DSD256), and HDMI multichannel. (A superior DIY M-Scaler?)

No worries! Putting a little low-power computer like this on top of the DAC did not affect noise level from the Sabaj DAC RCA/XLR outputs. Don't do this with more powerful computers obviously...

As expressed previously, my philosophy around computer audio is that given the speed of progress in computing technology, it really makes no sense to be building expensive and very fast computers as audio playback/processing devices. For me, if I have a general computer in an audio room, I would want to keep electrical noise low (ie. low power) while also targeting an ideally fanless solution. Inevitably, in less than 10 years, a powerful computer today would be very much obsolete. Monster machines would more than likely end up at the bottom of your closet with little value or interest left even if one wanted to sell. In fact, unless I'm doing a lot of 3D graphics, gaming, or editing videos, IMO progressively smaller, quieter, lower-power (let's aim for 10W or less with excellent performance!), and less expensive machines are where progress is heading for consumer-level general computing.

This is in part why on this blog, I've expressed much more interested in streamer systems like the Raspberry Pi devices or recently silent or very quiet MiniPCs (like the power-limited Beelink Mini S last week). As time goes on, one can judge objectively if technological progress and software updates like new DSP algorithms represent improvements in sound quality. If there are truly gains to be had, the machine can then be repurposed/replaced without feeling that one has wasted hundreds if not thousands of dollars.

At the end of the article last week, I spoke about how I reduced the fan speed and lowered the Mini S power limit to basically "silent" performance. Today, let's consider further what we can do with a machine like that in the audiophile sound room.

MEASUREMENTS / REVIEW: Topping D90SE DAC - Part I: The Hardware, Filters, MQA, and Jitter Performance

Today, let's start with what I'm aiming to be a 3-part review/discussion of the Topping D90SE DAC. I figure I'll take my time on this since things are busy around here and I hope not to purchase more hi-fi DACs going forward, so I might as well savor the moment. ;-) As a "more objective" audiophile who has a vision of aiming for "high fidelity" / "transparency", there is a clear target and end-point to what's needed from hardware performance especially with DACs.

Readers of this blog know that I've had a number of Topping devices reviewed here over the years. In fact, for any single brand of DACs, I think I've reviewed more Topping gear here than other brands. This, I believe, is a reflection of a brand that provides multiple products at price points with features that actually speak to me as a consumer interested in value which includes features, and price. I have certainly not been disappointed by overall quality to this point.

At a current retail price of around US$900, this is not an inexpensive model. As per most of my reviews, I bought this through usual retail channels so what I'm reporting on here is not any potentially specially-selected unit sent from the manufacturer.

Feature-wise, this is a DAC only but accepts multiple inputs - USB, HDMI-style I2S, S/PDIF Coax, S/PDIF TosLink, and wireless Bluetooth 5.0 (SBC, AAC, aptX [LL,HD], LDAC). While it has both unbalanced RCA and balanced XLR outputs, there is no headphone amp. For that, Topping recommends grabbing their top model the Topping A90 (~US$500).

What makes this DAC interesting in mid/late 2021 is that this is reputed to be the highest fidelity converter in the world (available to consumers) at this point in history, Topping's "flagship" device. 

Let's take a look...

Mega-taps upsampling: Remastero's Pan Galactic Gargle Blaster (PGGB) software. (Broadly on audiophile software & the noise boogeyman.)

This article came about after I received an E-mail from an audiophile friend who saw this Audiophile Style thread in praise of "math and magic". It links to a piece of software by a site called remastero, and the program itself is called "PGGB" (Pan Galactic Gargle Blaster), obviously referring to The Hitchhiker's Guide To The Galaxy with the main author named Zaphod Beeblebrox (who in the book is also the ex-president of the Galaxy). Cute, and of course the number "42" features prominently here and there.

In the past, we have talked about "audiophile" software that supposedly affect sound quality. Years ago, we talked about bit-perfect players (Windows, Mac) and really how "bit-perfect" is simply "bit-perfect" regardless of what software is used. We discussed questionable programs like JPLAY. Then there are the OS tweaks like Fidelizer. Neither JPLAY nor Fidelizer made any difference in my testing or listening.

That is not to say software doesn't make a difference at all. With the computing power we have these days, we can certainly perform highly precise filtering and DSD-PCM transcoding - like with HQPlayer. 

The idea with PGGB is that this is software that will take (in batch) various tracks you have and convert these to upsampled versions like 24/384 or 32/705.6 or even higher. In the process, applying very strong filtering (eg. on the order of >200M-taps sinc filter for some of the tests we'll run here, very impressive big number, right?). Furthermore, the website states that the software can apply settings for various levels of "transparency", apply HF noise filtering, uses noise shaping, adjusts gain monitoring for intersample overs, deal with convolution filters, and an apodizing setting. That's a bit of stuff so I won't promise that we'll hit on all these here. My intent is to at least have a good look at the foundation of the upsampling effect and the EQ function.

MEASUREMENTS: Roon 1.6 Upsampling Digital Filter Options & A Discussion on "Signal Path" Quality...

As discussed last month, I've started using Roon as my main music player for the sound room recently. Back in the days of Roon 1.2, many users performed upsampling using HQPlayer. While HQPlayer integration is still available (go to Settings --> Setup to access the installation option), since version 1.3, Roon has incorporated its own DSP samplerate conversion which I suspect would be completely adequate for the majority of users.

I was curious about the upsampling digital filter options available in Roon. If you look at the "Sample Rate Conversion" control panel, we see the four main "Sample Rate Conversion Filter" settings:

On the left panel, notice that Roon allows you to select the different DSP options and add various filters to the "chain" (left lower panel). "Headroom Management" is always available if needed which basically means you can set the amount of attenuation you want to use to prevent clipping while doing the DSP processes. Default setting is a very reasonable -3dB.

MEASUREMENTS: A look at the audio "ultra high-end" - ultrasonics! (And changes at Stereophile announced.)

In this blog installment, let's look at the "ultra high-end" of sound. Of course, I'm not talking about the audiophile "high-end" marketing term which is meaningless (beyond just another phrase for "expensive"). Rather, let's look at the frequency high-end, especially all the stuff our DACs can produce in the ultrasonic range!

If you've ever wondered, ultrasound devices used in medical imaging typically function at around 2MHz on the low end up to about 15MHz. However, the term "ultrasound" simply refers to wave
compression and rarefaction outside of the hearing range which by convention are those above 20kHz or so.

While the presence of ultrasonic content coming out of our DACs is not a surprise, what might be unclear or debatable is whether there is much of it and whether this then affects the "sound" of one's system. Rather than get bogged down in opinions, let's first have a look at what's in the ultrasonic frequencies coming out of DACs... Starting from facts, we can then perhaps come up with opinions.

MUSINGS / MEASUREMENTS: On why 24/96 sampling of the Blind Test of 16/44.1 devices is good enough. [On idealized filtering tests and "real life" music playback.]

As promised, I want to address a comment raised about the idea that the 24/96 recording using the RME ADI-2 Pro FS ADC was somehow inadequate for the 16/44.1 samples in the "Do digital audio players sound different?" blind test; that the 24/96 captures are somehow missing significant amounts of the output from the devices.

For reference, I've included the original comment here from Miska:

"It [the samples] just barely catches the first image band between 22.05 - 44.1 kHz and a little bit of next between 44.1 kHz and 48 kHz. One of the devices is clean though. The ADC anti-alias filter fixes rest by improving the reconstruction by removing further image frequencies. Strongest image for most current DACs is around 352.8 kHz.

Of course playback system then defines how much overlay of these come back again at different frequencies from the DAC, since it is now running at different rate family than the original source.

This doesn't really replicate real device playback performance, but at least it captures some apparent differences. So it gives kind of hint or shadow of how the device actually did."

Although the blind test is ongoing, I figure it's still good to talk about this while not revealing the devices used of course.

MEASUREMENTS: A Look At HQPlayer 3.25; Filtering, Dithering and DSD Conversion.

Over the years, no doubt many computer audio users have heard or perhaps tried using HQPlayer Desktop from Signalyst for music playback. While there is a GUI for playback as well as a sophisticated network transport architecture, it's the upsampling and PCM-to-DSD features that are the program's claim to fame. There has been a good amount of talk about the sound quality of the upsampling algorithms and some swear by the sonic differences the program makes.

While admittedly I don't follow HQPlayer chatter closely, I don't recall reading about specific settings and what they do other than opinions about the sound quality. In this way, HQPlayer has been a bit of a mystery and I know some folks have had difficulty getting it running over the years. It has certainly been on my list of items to look into. In fact, back in early 2017 when I wrote about Roon 1.2, I mentioned installing HQPlayer but never got around to actually writing about it.

Well, after all these years, and now with the requisite pieces of hardware at my disposal for a more thorough evaluation (reasonably fast CPU/GPU system, ADC capable of >192kHz, DAC capable of DSD512), it's time to have a peek "under the hood" at what it does...

NOS vs. Digital Filtering DACs: Exploring filtering turned off, implications, fidelity and subjective audibility. (Recent BorderPatrol DAC chatter...)

Hmmm... Non-OverSampled waveforms - "accurate", "high fidelity"?

The waveforms above were captured with my RME ADI-2 Pro FS ADC at 384kHz. As you can see, on the left side, we see the waveform from the TEAC UD-501 DAC with the digital filter turned off (Non-OverSampling - NOS mode). In the middle, we have the TEAC's "Sharp" oversampling filter engaged. And on the right, we see the same waveforms captured from an old 16-bit Philips TDA1543 x 4 DAC with chips in parallel (supposedly improves noise level), a multi-bit NOS chip, without any oversampling.

Here's the TDA1543 x 4 DAC board. Typically fed with a 12V DC power supply.

MUSINGS: Raspberry Pi 3 B+ "Touch" Optimizations; CRAAP Settings, and the "Extremus" Filter Setting.

Okay, let's have some fun with the Raspberry Pi 3 B+ motherboard, piCorePlayer, and my little "Touch" device I recently updated. As discussed before and in the picture shown above, these days, I have the inexpensive JustBoom Digi HAT (~US$40) board attached out the back for simple S/PDIF connectivity as well as USB when I need.

Today, let's hit a few topics related to the Pi 3 B+ setup above. Some little tweaks, things to get it working well, and fooling around with a piCorePlayer/SoX filter you can try for yourself without unnecessary hype nor expense...

Update: Raspberry Pi 3 B+ "Touch" Streamer, JustBoom Digi HAT, 3.0A power supply... And did they mess up Coltrane's "Both Directions At Once" in hi-res!?

Back in early 2017, I documented on the building of my little Pi "Touch" music streamer that I've been using over the last while for playback through my Logitech Media Server system at home. Over the last year, I've used this quite a bit for music playback and also some of the testing I've done on this blog.

As some of you may know, there has been an update of the Raspberry Pi 3 to the B+ model (~US$40). The upgrade isn't a major change to the existing Pi 3 board. However the new SOC, based on the Broadcom BCM2837B0 quad-core Cortex A53 (ARMv8, 64-bits) is slightly faster at 1.4GHz (1.2GHz previously, about 15-20% speed gain), and another upgrade is better ethernet speed which supports gigabit link but since it's still communicating through the USB2.0 port, will max out at ~300Mbps. In some benchmarks I've seen, though not true gigabit speed, the throughput is about 2x that of the previous Pi 3. For those interested in wireless connectivity, there are upgrades in the WiFi and Bluetooth departments as well.

Audiophile Myth #260: The Detestable Digital Filter Ringing and Real Music...

For the 2016 edition, go here.

In the audiophile world, there are many targets of angst and anxiety that "practitioners" of this hobby are well aware of.

For years we've been worried about the "dreaded jitter". However, we know that these days, with asynchronous interfaces like USB and ethernet, there's nothing to be concerned of. Sure, we can see jitter anomalies with old S/PDIF, but I doubt anyone should purposely not use the interface for fear of audible issues assuming otherwise decent gear. For years and still to this day, various "practitioners" of audiophilia hang on to beliefs around cables of all sorts; assuming normal hook-ups with decent quality cabling (and even with poor quality cables), we are typically hard pressed to find evidence of audible differences.

Then we have beliefs that bit-perfect streamers sound different, really? We also hear of esoterica like folks who think lossless file formats sound different, seriously? How about the folks who think that computer OS's make a difference, or software players sound markedly different (assuming it's all bitperfect of course and sent to the same DAC)...

Feel free to browse this blog for discussions and thoughts around countless other audiophile items of faith. Today, let's address the audiophile "myth" that has gained prominence among those trying to sell things and those that advertise said "things" over the last few years.

As per the title, today, let's explore this "myth" of the detested ringing with digital filtering and audio playback since I've been posting a bit of a series on this topic over the last little while.

MUSINGS: More fun with digital filters! Archimago's "Goldilocks" Intermediate Phase suggestion...

Today's post is a bit of a continuation from last time's look at the different types of upsampling anti-imaging playback filters using my Raspberry Pi 3 "Touch", piCorePlayer and SoX. As you can see from that discussion, across the audiophile equipment spectrum, manufacturers utilize all kinds of digital filter settings in their gear. Each company ends up choosing compromises between how much frequency roll-off, how much imaging, how much temporal/phasic anomaly each would accept. And of course no matter what a company chooses, there are ways of advertising the decision as "good"; whether it be on the basis of frequency spectral accuracy, temporal accuracy, or just claims from pure subjectivity - "it just sounds better"!

The end goal of audiophilia is a bit like the modern interpretation of Goldilocks (and the Three Bears)... We're all trying to figure out for ourselves what is "just right" as we wade through the commercial and mainstream audiophile literature, unofficial blogs and forums, mix-and-match speakers with amplifiers, try out different accessories perhaps, and the like. So too it seems with digital filters and all the variants attached to the DACs we buy.

Remember that the only reason we're even talking about this is because of that 44.1kHz (and to a lesser degree 48kHz) samplerate such that the Nyquist frequency is at 22.05kHz; relatively close to the usual 20kHz upper limit of hearing acuity that the younger ones among us might be able to perceive. This is literally the only reason for all the hand wringing and millions of spilt keystrokes over the years around filtering by audiophiles (the few who still obsess over this...)

These days, we essentially have 2 major options for filter "types" among the DACs out there... Linear phase (the default for most mainstream DACs, Chord) or Minimum phase (Apple, MQA, Pono) - pick your "poison" :-). Of course within each phasic variety we have different levels of steepness and allowance for ultrasonic imaging. We intuitively know that due to the biological phenomenon of auditory masking, maximum phase (where the group delay is pushed forward so "pre-ringing" is accentuated) is not desirable. But is there another choice?

Yes, there is of course... We can try to figure out a "just right" state with intermediate phase settings. Accepting that maybe there's some value to ensuring that pre-ringing isn't an issue even with some of the worst audio recordings out there, while maintaining awesome frequency and temporal accuracy - let me show you my choice for the filter that I listen to daily with the Pi 3 streamer...

HOWTO / MUSINGS: Playing with Digital Filtering - Impulse Responses and Frequency Effects resembling Chord, Old-Skool Meridian, MQA, NOS...

Remember folks... With anti-imaging filtering applied to digital playback, there are no gross "stairsteps" in the oversampled/upsampled analogue output. Regardless of what you might see from misleading audio ads trying to pigeon-hole "digital". The exception being non-oversampling DACs that specifically choose to go down the stair-step path.

As you may know, I've been using a Raspberry Pi streamer with LMS over the last year for most of my listening. As described in that Pi "HOWTO" article, I've been using the piCorePlayer software to stream.

One of the nice features of piCorePlayer is that it allows easy control of upsampling by using SoX; in the "Advanced" settings is the ability to pass along instructions to control the digital filtering used. Combined with a good DAC that accepts high sample rate PCM, we can try listening for ourselves and gain an appreciation of the differences that digital filtering make. The cool part of this is that it allows us to "emulate" the kinds of filters used in the audiophile world.

In this post, let's play with some of the parameters and show the kinds of impulse responses and "digital filter composite" graphs we can achieve...

MEASUREMENTS: The AudioQuest Dragonfly Black MQA filter set...

Good movie from the 90's!

Just a quick note while we're doing the MQA blind test!

A few weeks ago I posted on the Dragonfly Black and the MQA filter impulse responses. I noted that based on Måns Rullgård's work, although there are potentially 32 filters available when "Rendering" with upsampling to 192+kHz, I only showed 0-8 in that previous post.

Well, using the same kind of impulse response diagram previously (captured using my Focusrite Forte 24/192 ADC), here's the AudioQuest Dragonfly Black "full monty" of impulses...

MEASUREMENTS: MQA Filters on Mytek Brooklyn & Thoughts on USB Doohickeys

Hey folks... It's summer and I'm in and out of town doing a bit of R&R mixed with the day job.

I sent my friend with the Mytek Brooklyn and Fireface ADC a package of the MQA "Render" tagged files for testing. As you can see above, that's what the MQA filters look like with the Mytek Brooklyn serving as "Renderer" for the impulse responses. If you look at the top row, we see that the Brooklyn's standard digital filter at 24/96 is the same as what MQA uses for playing back 96kHz "original" unfolded PCM. This is different from the Dragonfly Black DAC presented a couple weeks back where the standard filter at 24/96 is a much sharper one and would shift to the weaker MQA filter only when it detects the MQA Render tagged data.

MEASUREMENTS: AudioQuest Dragonfly Black 1.5 - PART 2 (On "MQA Rendering")

AudioQuest Dragonfly Black - "rendering" MQA playback. Dragonfly logo turns purplish...

For those keeping track, the story of MQA has slowly evolved over the last few years. Initially, it was supposed to be just a technology incorporated into DACs that would take an encoded file (typically 24/44 or 24/48) and almost "magically" transform this data into the "original" high resolution (equivalent to 24/192+) sound while maintaining standard PCM playback compatibility. This was the case with the first products like the Meridian Explorer 2. Over the years, it appears that concessions have been made. Software decoding rather suddenly was introduced in early 2017 with TIDAL's support for MQA; branded as "MQA Core". At around the same time, DACs were classified as either "MQA Decoders" for components that can handle these 24/44 or 24/48 files start to finish (Meridian, Mytek DACs), and "MQA Renderers" which require the computer software perform initial "unfolding" to high-res samplerates (88 or 96kHz) followed by some kind of final processing to the sound performed by the DAC presumably to allow the sound to be more "accurate" to the studio production, or "authentic" (whatever that means).

With the recent firmware upgrade to version 1.06, the Dragonfly Black DAC now belongs to this latter category of "Renderer". As far as I am aware, it is the first of these kinds of devices to be released without "Decoding" capability.

MUSINGS: Discussion on the MQA filter (and filters in general)... [Update: Including a look at the classic "Apodizing" filter.]

Here's an interesting comment from the post last week...

CBee 7 February 2017 at 04:47

Excellent article but I have one query. On Sound on Sound they say "MQA claim that the total impulse-response duration is reduced to about 50µs (from around 500µs for a standard 24/192 system), and that the leading-edge uncertainty of transients comes down to just 4µs (from roughly 250µs in a 24/192 system)." In that case wouldn't you need an ADC with higher resolution than the RME Fireface 802 in order to see any real differences between the Reference and Hardware MQA decode?

As I said... Dammit CBee! Now you've made me post another blog entry on MQA :-).

MEASUREMENTS: HiFiBerry DAC+ Pro, PCM5122 digital filters & further thoughts on filtering...

A summary slide from this presentation slide set. Just a reminder of the general desirability of digital filters providing higher accuracy compared to analogue...

Thanks to Jim Ambras in the discussion of the measurements for the Raspberry Pi 3 & HiFiBerry DAC+ Pro for pointing me to some settings available! So in the "Wait! There's more..." department, there are some "advanced" settings available in Alsamixer for the DAC+ Pro I2S HAT board you might wish to experiment with. Let us have a look at the different digital filters built into this very affordable DAC and how it changes the sound objectively...

MUSINGS: Digital Interpolation Filters and Ringing (plus other Nyquist discussions and "proof" of High-Resolution Audio audibility)

A couple weeks ago, Whackamus posed this interesting comment and question which I thought would be a good topic to discuss and explore in greater detail and with some examples/samples:

"I've been reading your blog for years. Or for almost four years, at any rate. I have to thank you for doing what you do. I've likewise always wanted to ask you a question, too, but I don't know how the bleep to to contact you. In any case, since I've been fretting over it afresh, I thought I'd just post it here. If you ever do decide to get to/address it, that'd be great. If not -- hey, no sweat. :)

In any case, I read the following (tonight) on the Stereophile forums:

"I personally think that MQA has some noble goals, in terms of getting as close to the original master as possible, but I think that is far less important than the elimination of the damaging pre-ringing distortion. This has been the bane of digital playback for 30 years, and over-sampling and various filter techniques have tried to deal with it, with limited success."

I won't say that I've never heard ringing -- because I probably have -- but I will say that I've never explicitly said: "Aha! Eureka! Thar be ringing!" Because -- outside of maybe a blurring during transients? -- I have no idea what it sounds like. But my question is less about MY having heard ringing than about the AUDIBILITY of ringing -- pre, post, or otherwise. In a quality DAC (which I've got to assume most of the folks posting on Stereophile.com have access to), how audible are ringing effects? Or, rather, how COMMON are they? I kind of imagine that the Meitners, Lavrys, Levinsons, Stuarts, etc. of the audio world take great care to minimize (pre-/post-)ringing effects and to eliminate ringing in the audible realm. I likewise imagine that both such things are doable, inasmuch as most of us have been enjoying digital audio for decades now. But the Stereophile poster makes it seem as if ringing is the apodeictic bane of digital audio. What am I missing?"