HQPlayer Impressions and Settings Rolling Thread
Apr 24, 2024 at 7:32 PM Post #1,156 of 1,340
Great! If you are using WiFi, the new NAA v5 protocol may be beneficial compared to the older v4 one. NAA OS provides v5 while I believe Red OS provides v4. I don't know what RopieeeXL provides. This mostly as heads up in case you encounter dropouts or something like that.
Thank you, I will check it out. A friend of mine with HQPlayer suggested I request to you adding functionality to select the network adapter that HQPlayer uses; certainly it can get confusing with VPNs and similar tools installed. I don’t know how complex or not that would be, but I can see it’s potential in saving some trouble.

Thanks again for your help!
 
Apr 24, 2024 at 7:36 PM Post #1,157 of 1,340
Thank you, I will check it out. A friend of mine with HQPlayer suggested I request to you adding functionality to select the network adapter that HQPlayer uses; certainly it can get confusing with VPNs and similar tools installed. I don’t know how complex or not that would be, but I can see it’s potential in saving some trouble.

Thanks again for your help!
It is already available to select NAA. I have Hqplayer select any one of my 4 NAA in 4 different systems to 4 different DACs in the same local network.
 
Apr 24, 2024 at 11:12 PM Post #1,158 of 1,340
If you are using WiFi, the new NAA v5 protocol may be beneficial compared to the older v4 one.
How it is in the case of mixed type of LAN connection? For example wireless from HQPlayer computer but wired to NAA? Is v5 beneficial in that case?
 
Apr 25, 2024 at 3:34 AM Post #1,159 of 1,340
With apodizing filters you can replace/reduce the ringing with less disturbing and possibly shorter one.
Does this have a potential beneficial effect even if the apodizing counter is at zero for a particular recording, or is it only useful when that counter indicates unappreciable tally?
 
Apr 25, 2024 at 8:40 AM Post #1,160 of 1,340
What looks like "perfect" filter (if we talk about steepness) from frequency domain point of view is absolutely opposite from time domain point of view. Because these are related through inverse 1/x relationship.

Likewise, in mathematically related aspect, when you do spectrogram with FFT (like in HQPlayer metering view) you need to choose transform length. When you make the transform longer, you gain more frequency resolution (in digital filters this means steeper roll-off), but your time resolution to detect changes suffers. You need to choose suitable balance between the two.

This time-frequency uncertainty is called Fourier uncertainty principle. Also remember that hearing can beat this uncertainty principle. See also here.
Yeah I think I have a bit of a grasp of what the uncertaity principle tells us. It also states that a band-limited signal must necessarily be infinite in length and that's why our recordings are not perfectly band-limited (and why we need short filters in the first place instead of super-long sinc filters).

The inverse 1/x relationship would make it seem like the perfect brickwall filter would be infinitely bad in the time domain since it rings infinitely long, and perfect in the frequency domain.
However since the perfect brickwall filter will perfectly reconstruct a sampled bandlimited signal, apparently this infinite badness in the time domain is of no consequence to bandlimited signals. (this is actually the same thing as saying a reconstruction filter will not ring when fed a bandlimited pulse, now that I think of it).
Which is I guess counterintuitive.

Also, since Currawong came into this thread asking about filter ringing (and stating under normal circumstances a dac filter doesn't ring) and kennyb123 talking about PGGB (which in this myths page https://www.remastero.com/faq-plus.html#Myths takes aim at the HQPlayer philosophy that digital filtering is a compromise between time and frequency domain) I have a need to figure out who is "right".

I did a little thought experiment where we'd have to assume two different types of recordings:

- A recording of an instrument with no frequency content above 20khz. No need to bandlimit this signal when putting it through an adc since it is inherently bandlimited and no ringing will get baked into the recording because of the adc filter.

- A recording of an instrument that had frequency content above 20khz. This recording will contain all the frequencies of the instrument below 20khz, plus ringing at the adc filter frequency where the higher frequencies had to be filtered out.

- Note that both these recordings will still not be bandlimited (remember, bandlimited signals don't exist in the real world). So both of them will have content above 20khz. What this looks like or where it comes from I don't know, I still have trouble wrapping my head around this fact. See this Wikipedia excerpt:
1714048463581.png


Now what happens if we try to reconstruct both these recordings with a perfect brickwall filter?

-In case one we get the exact original signal without ringing, plus ringing artifacts for all the out-of-band parts of the recording.
-In case two we get the exact original signal after bandlimiting, plus all the ringing from the adc filter, plus ringing artifacts for all the out-of-band parts of the recording.

A short apodizing filter would in case two replace the adc filter ringing with its own presumably better ringing, and would ring less as a result of the out-of-band parts.

From this little thought experiment I draw three conclusions:
1. The HQPlayer view and the Chord/PGGB view of signal reconstruction do not have conflicts when it comes to the sampling theorem.
2. The HQPlayer view is that adc ringing + out-of-band signal ringing is quite detrimental to the quality of the reconstructed signal.
3. The Chord/PGGB view is that adc ringing + out-of-band signal ringing is completely inconsequential for the reconstructed signal as humans cannot hear the ringing and there is barely any out-of-band signal energy in recordings anyway.

Now this became quite a long post but it pains me that our audiophile community seems yet again to have devolved into two camps and I naively would like to figure out how we can prove which view is right, especially since this seems to be one of the rare audiophile discussions that might actually be resolved by scientific means.

Anyone with any ideas on how we could visualise the time-domain badness of the ringing caused by out-of-band components of the signal?
My ears tell me short filters sound quite different compared to long ones so the badness must be significant, but unfortunately listening impressions are not the same as evidence :sweat_smile:
 
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Apr 25, 2024 at 8:50 AM Post #1,161 of 1,340
How it is in the case of mixed type of LAN connection? For example wireless from HQPlayer computer but wired to NAA? Is v5 beneficial in that case?

If there is wireless on the path, then it is beneficial. For pure Ethernet it may help in some cases, but I don't expect so much difference. The protocol is somewhat different now though. Also to support the new features it has.
 
Apr 25, 2024 at 9:30 AM Post #1,162 of 1,340
Does this have a potential beneficial effect even if the apodizing counter is at zero for a particular recording, or is it only useful when that counter indicates unappreciable tally?

It is less likely to have effect on that when the counter stays zero. The counter is not absolute, it has quite a bit of safety margin to avoid misdetections and such.

- A recording of an instrument with no frequency content above 20khz. No need to bandlimit this signal when putting it through an adc since it is inherently bandlimited and no ringing will get baked into the recording because of the adc filter.

- A recording of an instrument that had frequency content above 20khz. This recording will contain all the frequencies of the instrument below 20khz, plus ringing at the adc filter frequency where the higher frequencies had to be filtered out.

This is the case yes.

Although your DAC side will also ring in case of digital clipping, even if the source signal didn't otherwise contain > 20 kHz. As we know, digital clipping is not at all unusual thing with RedBook content.

See this Wikipedia excerpt:
1714048463581.png

This is what I've been talking about.

-In case two we get the exact original signal after bandlimiting, plus all the ringing from the adc filter (this could have been replaced with an apodizing filter)

Yes...

view is that adc ringing + out-of-band signal ringing is completely inconsequential for the reconstructed signal as humans cannot hear the ringing

It would also mean it doesn't matter what oversampling filter you use because there wouldn't be any difference in those, since they only affect outcome above audio band.

Yet you have usually choice of different filters, even inside DAC chips. There could be a reason?

and there is barely any out-of-band signal energy in recordings anyway

I just don't understand where this is coming from. Maybe people have not looked at RedBook content and HiRes content. Or listen only to piano music (max 16 kHz)?

Screenshot from 2024-04-25 16-05-24.png
 
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Apr 25, 2024 at 9:59 AM Post #1,164 of 1,340
It would also mean it doesn't matter what oversampling filter you use because there wouldn't be any difference in those, since they only affect outcome above audio band.

Yet you have usually choice of different filters, even inside DAC chips. There could be a reason?

I just don't understand where this is coming from. Maybe people have not looked at RedBook content and HiRes content. Or listen only to piano music (max 16 kHz)?
Don't shoot the messenger :sweat_smile: I just try to rephrase what I read. My ears tell me filters make a significant impact in the audible band!

Interesting that even this sonically perfect Daft Punk recording contains ultrasonic junk, you can clearly see that the strong transients are mirrored around 42 Khz!
 
Apr 25, 2024 at 10:23 AM Post #1,165 of 1,340
Interesting that even this sonically perfect Daft Punk recording contains ultrasonic junk, you can clearly see that the strong transients are mirrored around 42 Khz!

A bit of aliasing from above 44.1 kHz. And a bit of SMPS/CRT whine at 29 kHz.
 
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Apr 25, 2024 at 11:00 AM Post #1,168 of 1,340
Step response. A good filter will have minimal overshoot and ringing should decay as quick as possible.
Thanks for responding!
The step response is the response to an infinite bandwidth signal so that doesn't exactly demonstrate the effects in real music i'm afraid.

I was thinking more along the lines of taking a hires 196khz recording, decimating it to 44.1 khz using a filter similar to an adc decimation filter, and then plotting the results of upsampling it again with poly-sinc-short-mp vs sinc-L. What effects would be observable in a time vs. amplitude plot for example?

I know Archimago did a similar experiment here;
https://archimago.blogspot.com/2018/01/audiophile-myth-260-detestable-digital.html

But his results didn't seem to advocate for using a short minimum phase filter unfortunately.
 
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Apr 25, 2024 at 12:39 PM Post #1,169 of 1,340
advocate for using a short minimum phase filter unfortunately.
Maybe I missed it but who is advocating using a minimum phase filters?

Probably a totally separate topic but the convolution fir filters I create are linear phase that preserve the time domain and correct for issues that were created by minimum phase phenomenon (Room reflections, frequency response anomalies, etc. etc)
The programs I use also have ringing compensation built in.
That time and phase correction ability is what seperates the men from the boys as far as state of the art room correction is concerned.
 
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Apr 25, 2024 at 1:36 PM Post #1,170 of 1,340
Maybe I missed it but who is advocating using a minimum phase filters?

Probably a totally separate topic but the convolution fir filters I create are linear phase that preserve the time domain and correct for issues that were created by minimum phase phenomenon (Room reflections, frequency response anomalies, etc. etc)
The programs I use also have ringing compensation built in.
That time and phase correction ability is what seperates the men from the boys as far as state of the art room correction is concerned.
Jussi considers poly-sinc-short-mp the ideal filter for music containing strong transients such as pop and rock as far as I know.

In my experience minimum phase filters indeed improve transients but they muddy spatial cues (sensitive to time domain distortion). However most pop music doesn't contain physical space to begin with (half the instruments may even be electronic samples) so not a lot is muddied.

You seem to know your stuff, do you agree that for headphones linear phase is more important since they are linear phase devices with a single driver while for speakers minimum phase is less damaging since they are multi-driver with minimum phase crossover filters and room reflections?
I've heard someone mention that and i've been wondering about it.
 

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