Are there any alternatives to HQPlayer?
Aug 22, 2023 at 12:06 PM Post #31 of 38
@Scarydog

Have you tried NAA OS for Holo Red? It's NAA only, but works better than standard. I use mine with this software and its most reliable endpoint that I ever had.

As for the rest, you'd better ask developer. He is very supportive person. You can write to Audiophile Style forum. HQPlayer's author nickname is Miska.

P.S. Check if you picked right "USB out" in NAA settings. There are two options there. One is for Red's USB out and another is for its DDC output (AES, I2S and other), but NAA sees them both as "USB".

P.P.S. For Holo May/Spring in PCM mode DAC bits should be set to 20 for best linearity.
 
Aug 22, 2023 at 5:29 PM Post #32 of 38
@Scarydog

Have you tried NAA OS for Holo Red? It's NAA only, but works better than standard. I use mine with this software and its most reliable endpoint that I ever had.

As for the rest, you'd better ask developer. He is very supportive person. You can write to Audiophile Style forum. HQPlayer's author nickname is Miska.

P.S. Check if you picked right "USB out" in NAA settings. There are two options there. One is for Red's USB out and another is for its DDC output (AES, I2S and other), but NAA sees them both as "USB".

P.P.S. For Holo May/Spring in PCM mode DAC bits should be set to 20 for best linearity.
Thanks! That is very helpful - I will give it a try.
 
Aug 28, 2023 at 5:48 AM Post #33 of 38
But because of how processing intensive it is, it's not intended to run realtime like HQPlayer.
I don't think it is the main reason. It installs the Matlab runtime while installing, meaning it is still using Matlab implementations of some functions which are not implemented natively and optimised for the PC and specific application. This is independent of how useful it is or if it should do real time processing. Probably the developer is good under Matlab but not so with optimised programming - which is what I often see. Having both is quite seldom. But I still would expect a software that sells for a grand (independent of the claims) to be well optimised for the target it is running.
Indeed it was invented for a reason and as the Wikipedia article explains, Noise Shaping changes the spectral shape of the resultant noise during dithering, hence why it’s called “Noise Shaped Dither”. However, that is the only noise that Noise-Shaping affects. So if for example a mastering engineer applies dither or noise shaped dither, that noise is now part of the recording. A subsequent noise-shaped dither process, say in the DAC, only shapes the noise of the dither the DAC is applying, it does not noise-shape the dither applied by the mastering engineer or the dither applied during AD conversion or indeed any other noise that’s already part of the recording.

G
...and I am yet to see any mixing / mastering engineer take any of the audiophile claims seriously. The original recordings are still often 44.1/48 kHz 16 bits, or max 96 kHz 24 bits. Then the files go through multiple layers of DSP with a maximum of double precision floats (meaning 64 bit floats), and then they end up on our listening gadgets.
I have played around with HQPlayer a few times and finally bit the bullet and bought a copy 6 months ago. I have repurposed an older Mac Mini now running linux and HQplayer Desktop.

It works fine, I can upsample PCM ok, not really enough grunt to upsample to DSD but that’s fine.

But I have a coupe of issues with HQPlayer.
The interface is terrible, really poor feedback and some really quirky bugs. I saw that new version has been released with a new interface. I downloaded it and tried it, it was better so I thought I would buy it. Unfortunatley the price to upgrade is nearly as high as the original purchase price, which I already thought was not great value.

The only alternative I can think of is Roon. It doesnt sound as good, but is certainly much easier to use (and much less expensive)

I have had a bit of a search for hardware solutions but can’t find quite what I am after.

I’d be grateful for any suggestions for software or hardware upsamplers.
You can also try the HQPlayer Pro, to see if you really hear any audible difference. Just download the demo version and convert some of your files.

https://www.signalyst.com/professional.html

...or you can also try this one.

https://samplerateconverter.com/
 
Aug 28, 2023 at 7:36 AM Post #34 of 38
I don't think it is the main reason.
It is the main reason. A real-time version is actually available as a foobar plug in but you can't process to the same level of precision as the standalone version.

When you're doing processing that is equivalent to hundreds of millions or billions of filter coefficients at 192 bit accuracy, I don't think any amount of optimisation is going to allow that to work real-time. Even just the speed at which data can move into RAM might even limit that in some instances id imagine but could be wrong.
 
Aug 28, 2023 at 9:28 AM Post #35 of 38
It is the main reason. A real-time version is actually available as a foobar plug in but you can't process to the same level of precision as the standalone version.

When you're doing processing that is equivalent to hundreds of millions or billions of filter coefficients at 192 bit accuracy, I don't think any amount of optimisation is going to allow that to work real-time. Even just the speed at which data can move into RAM might even limit that in some instances id imagine but could be wrong.
Oooooh, OK. Billions. So it has gone soooo wild, huh... That explains why 7 hours are needed for 3 pieces here. That was beyond my imagination that someone would dare to do that, even more someone would claim be able to hear that. Still, it is using the Matlab runtime, it is not there for no reason. But Enjoy your 256 bit floats with billions of coefficients.
 
Aug 29, 2023 at 3:22 AM Post #36 of 38
...and I am yet to see any mixing / mastering engineer take any of the audiophile claims seriously.
It’s been going on for over 60 years now. As technology improved and artefacts of the various pieces of equipment in the chain were forced ever further below the thresholds of audibility with ever cheaper solutions, the audiophile industry had no option but to take problems that were solved 30-40 years ago (or simply invent fake problems that never existed), imply/state they’re current audible problems and then provide solutions to these fake problems at audiophile prices. Meanwhile, the pro-audio industry (mix/mastering engineers, etc.) either just roll their eyes at the ever more outlandish audiophile claims or far more commonly these days, simply ignore/treat the audiophile world as they would the climate deniers, flat-earthers, anti-vaxxers and horoscope communities.
The original recordings are still often 44.1/48 kHz 16 bits, or max 96 kHz 24 bits.
To be fair, commercial recordings are pretty much never made at 16bits anymore. 44.1/48kHz 24bit has been and still is far and away the most common for around two decades or so. The rest of your post is entirely correct though. 64bit float has been the standard mix and mastering environment for over a decade (and 32bit float or 48bit fixed for more than a decade before that).
When you're doing processing that is equivalent to hundreds of millions or billions of filter coefficients at 192 bit accuracy, I don't think any amount of optimisation is going to allow that to work real-time.
Yet even an iPhone 12 could do 11 trillion operations per second in real-time. Billions of filter coefficients is ridiculous when a few thousand is already beyond audibility and 192bit accuracy is just as ridiculous when the input file is 16bit or at most 24bit. But more is always better in the audiophile world, even when it’s actually worse, because audiophiles are willing to pay for the privilege of worse (or no better).
Even just the speed at which data can move into RAM might even limit that in some instances id imagine but could be wrong.
Stereo 16/44 requires 1.4 megabits per second and even 192/24 only uses about 9.2 megabits per second. The HD to RAM data transmission speed of say a Mac Studio is around 7,000 megabits per second and the RAM-CPU data bandwidth is 800 gigabits per second! How could anyone even imagine there might be a data bandwidth problem?

G
 
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Aug 29, 2023 at 8:23 AM Post #37 of 38
Yet even an iPhone 12 could do 11 trillion operations per second in real-time. Billions of filter coefficients is ridiculous when a few thousand is already beyond audibility and 192bit accuracy is just as ridiculous when the input file is 16bit or at most 24bit. But more is always better in the audiophile world, even when it’s actually worse, because audiophiles are willing to pay for the privilege of worse (or no better).
A single operation cannot be directly equated to a filter coefficient. That isn't how things work.

When you process a track like this, you are processing the ENTIRE track at once, with those hundreds of millions of filter coefficients, hence why PGGB will utilise in some instances a couple hundred gigabytes of RAM if you have it available. It isn't as simple as doing one sample and then moving on to the next.

You can do block processing, but that then limits how many coefficients you can use and has other limitations.

In regards to the 192 bit precision, this is not about outputting a 192bit file, but doing the maths itself with 192 bit precision, which surprisingly enough CAN affect the value of 24 bit samples. The bit depth of the file and the precision of the maths are separate and I'm sure you're aware of this already. For similar reasons as to why with a 16 bit ADC you can still look lower than -96dB with FFT gain etc.

For example here's a test showing some small signal content at 44.1khz upsampled to 705.6khz using 64 bit precision and output to a 24 bit file.

1693311204608.png


You can see the noise shaper is effective to about -380dB here. So, very good (64 bit precision is already a lot of course). But to demonstrate that regardless of the fact that we are only outputting to a 24 bit file we can still affect the content:

256 bit precision, still outputting to 24 bit file, and look at the result:
1693311296479.png


Now we have a noise shaper accurate to around -600dB. And we will be able to provide more accurate phase and amplitude accuracy for the reconstructed content. Whether that is audible/useful is an entirely separate debate and this isn't the thread to have that discussion (nor do I really wish to have that debate as it's pretty clear where we both stand and that's unlikely to change for the moment).
My point is simply that the higher bit precision is not about the file output, but about how the maths itself is done and how that then affects the output of even a 24 bit file.

Lastly in regards to the mentions of PGGB using MatLab and 'being unoptimised', the dev has added a post to the site discussing this. The TLDR is that the core library is fully optimized C++ which uses assembly level optimization, Matlab is a shell that provides a user interface and calls the C++ PGGB library. The reason PGGB takes so long to process a file is simply because processing tracks with millions of samples with hundreds of millions of filter coefficients at high levels of mathematical precision is just a very compute intensive task and there isn't a way around that.

1693311734141.png
 
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Aug 29, 2023 at 10:54 AM Post #38 of 38
So, very good (64 bit precision is already a lot of course).
It’s not “very good”, it’s very bad! We use 64bit because that allows us to mix over a 1,000 channels of audio with thousands of plugins and rounding errors to still be kept way below audibility. How many 100,000’s of channels and how many millions of plugins are audiophiles using?

The -380dB or even -600dB noise floor level is even more ludicrous. -144dB is roughly the sound of 2 hydrogen atoms colliding and you’re talking about a noise floor a trillion times quieter than that. You do know that the noise floor of just air and no other sound is -23dBSPL and that 380dB higher than that is around 10,000 times more sound pressure than the biggest atomic bomb ever exploded? How do audiophiles swallow this ridiculous marketing without falling over laughing?

G
 

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