Using the MeLE Mini PC for Upsampling/Filtering - HQPlayer Desktop 4 streaming from Roon. [Going forward, AVX2 needed for HQPlayer + Roon... For some reason.]

Hey folks, a few weeks ago, I mentioned that I've got the little MeLE Quieter2Q Windows 10/11 computer in my sound room these days used as a low-power (<10W) Roon streamer capable of 2.0 and 5.1 multichannel. The computer is relatively powerful for just an audio device handling Roon streams; 8GB DDR4 RAM, a 2.0+GHz turbo Celeron J4125 quad-core CPU is certainly much more than needed just to basically get data from the gigabit ethernet and shove it to USB or HDMI for playback!

It works fantastically also as a little fanless HTPC for movies with 4K video and audio bitstreaming. A tip: K-Lite Codec Pack works very well, it can even play MKV/MKA files with lossless TrueHD-Atmos like the 2019 Beatles Abbey Road BluRay rip which I know will trip up some HDMI hardware.

I mentioned in the previous article that it might be fun to consider utilizing the processing power of the little fanless box for audio. For many Roon users, one way of taking advantage of processing power is by using HQPlayer from Signalyst for high-quality upsampling with digital filtering, and possibly even PCM-to-DSD conversion. 

Upsampling: Native DAC Playback, and SoX PCM/DSD upsampling of 1kHz signal. (And the "Beatles: Get Back" documentary, "As the artist intended"?)

Notice one of the waveforms NOS.

A few weeks ago on this blog in a comment discussion, Bennet / Dtmer Hk talked about showing what it looks like when upsampling a 1kHz 16/44.1 signal in PCM side-by-side with DSD upsampling.

Sure, no problem! We can have a quick peek at the 1kHz sine tone from a couple of DACs for comparisons between direct playback with built-in filtering, upsampled in high quality with PCM SoX, and then using high quality DSD conversion with SoX-DSD.

Plus, let's also have a look at the recent Beatles: Get Back documentary series and some thoughts which I think relate not just to the "music lover" which I hope is in all of us, but also to the "hardware audiophile" side of this hobby.

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.

GUEST POST: Why We Should Use Software Volume Control / Management by Bennet Ng. (Plus discussions on resampling options, true peaks, etc...)

I received an invitation from Archimago to write something about volume control. While I am of the opinion that digital and analog volume controls can coexist to achieve an ideal gain stage, this article is mainly about PCM digital volume control.

The basic conclusion we can say regarding digital volume control is that as long as the playback device has higher bit-depth than the file source, it is possible to losslessly reduce the volume of a file until the playback device's bit-depth limit is reached. For example, with an ideal 24-bit device, it is possible to playback a 16-bit file 48dB lower without losing quality, because one bit has about 6dB of dynamic range (the exact formula of bitdepth and dynamic range is 6.02*n-bits + 1.76).

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 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...

MUSINGS / MEASUREMENTS: A look at Intersample Peaks and Overload Tolerance... (RME ADI-2 Pro FS, TEAC UD-501, Oppo UDP-205)

I suspect that many audiophiles were introduced to the topic of intersample peaks (ISP) when Benchmark first released their DAC2 HGC which advertised and discussed the provision of +3.5dB overhead for digital conversion back in late 2012. Over the years they've discussed this in their blog such as this post that came out to discuss the technical bits of the Sabre DAC and DAC2 in particular.

Remember however that ISPs and the potential for intersample overloading have been discussed in the technical literature for years. For a good review of the topic, have a look at this paper from Nielsen and Lund "Overload in Signal Conversion" presented at the AES Conference in 2003. Because almost all modern DACs utilize oversampling filters, the internal interpolation creates extra "points" between samples. For loud music (common since the late 1990's!) with samples approaching the 0dBFS level if not already clipping, it's likely that many of these intermediate points created will be above 0dBFS. Even in 2003, the paper listed a number of albums and showed that quite a few have "hot spots" where the "true peak amplitude", after going through an interpolator would be above 0dBFS. (A NOS DAC doesn't do digital intersample interpolation but could still overload with analogue filters [as per Mans' comment].)

SET-UP: Low Power Linux Audio Player (ODROID-C2 & Volumio 2)

As you know from a few weeks back, I got myself one of these little single-board computers (SBC) - the Hardkernel ODROID-C2 - and posted a PREVIEW on it. As described in that article, I had flashed a copy of Volumio 2 "Release Candidate" (0.861RC1) to my SDHC card and have been streaming music to my TEAC UD-501 DAC in the music room for the last while.

Originally, I was going to publish both the contents in this post as well as measurements but it quickly became obvious that this was going to be too unwieldy (plus the day job got very busy)! As such, let me talk about the basics of the setup here today along with a few changes you might want to try if you're streaming from one of these Linux-based network machines. Then later, we'll get to the measurements...

Note that although I'll obviously be specifically addressing the ODROID-C2 I have here, the software is portable and at present, the good folks at Volumio have software images for many low-power computers including: Raspberry Pi, Beaglebone Black, Cubox, ODROID-C1+, etc... Just check their site to see if your machine is supported with a ready-to-use OS/app image. I've been informed that RuneAudio will also have ODROID-C2 support ahead (maybe as early as this weekend), so keep an eye on that one as well!

MEASUREMENTS: Apple Mac OS X ("Yosemite") Software Audio Upsampling

Well... You knew this was coming, right? After looking at Linux, then Windows, what about all the hub bub with Mac OS X? Depending on which forum one visits, there's often talk about Mac's being "better" as audiophile sources than PCs.

As readers here know by now, I think sound quality these days with a decent modern digital interface is primarily a function of the quality of one's DAC. So long as the digital source device is decent (not a high bar to cross by any means!) with bitperfect results and absent of any timeout errors, or extreme noise pollution, it's just fine. And one can verify the quality with objective measurements of course.

The question being asked today is, compared to the software upsampling in Windows and Linux, how well does the Mac OS X software upsampling measure up?

MEASUREMENTS: Windows 10 "Audio Stack" / DirectSound Upsampling

I know, I know... Windows audio mixer sucks... (From the perspective of perfectionist audiophiles.)

But having just done some measurements with Linux and PulseAudio with some of the upsampling algorithms, I wondered just how well the default Windows 10 audio mixer performed as an upsampler...

MEASUREMENTS: A Look at Linux Audio (ALSA, PulseAudio)

What is that you ask? Well, it is the slowest computer in my house :-). It's a "vintage" HP Mini 110 from circa 2009. I decided recently to stick Linux on there just to see how well it ran... Here's the System Information:

As you see, I've got Ubuntu 14.04 LTS (updates current as of Sept 2015) loaded up on the old machine which has been sitting around unused for a few years. It's one of the early Intel Atom N270 CPU's (32-bit, single-core hyperthreading, 1.6GHz), 2GB DDR2 RAM. Slow 16GB SSD as the main drive. Whole OS loaded up on the tiny storage, no problem. Linux runs fine on the machine but it's obviously not fast, but works for light web surfing, E-mail, word processing... Now, how about for audio?

LIST: Suspected 44 or 48kHz PCM upsampled SACDs.

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

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

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

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